Linux-audio-user September 2005

linux-audio-user@lists.linuxaudio.org

157 participants
172 discussions

[linux-audio-user] [OT] Audio Libre pdf file problems

by Juhana Sadeharju

Hello. The files lud50-Audio_Libre.pdf lud51-Audio_Libre.pdf at http://linuxaudio.org/en/press/ are shown without text. xpdf says: Error: Couldn't find a font for the 'Adobe-Identity' character collection How I could say "I don't have Adobe-Identity, please use This"? I have xpdf 1.0. Does xpdf 2.0 fix the problem? What would be most portable font which Libre people could use instead of Adobe fonts? BTW, I just downloaded the front webpages of all audio and graphics at sourceforge (16377 in total). Many of the front pages works the same way as Audio Libre documents. But unfortunately many of the front pages does not explane what the software is and does; only a changelog (named as "news") and installation guide are provided. What your software's front page looks like? Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

18 years, 8 months

[linux-audio-user] Re: Creative Live! Drive midi ports?

by Juhana Sadeharju

>From: Erik Steffl <steffl(a)bigfoot.com> > >website), not only that but there was only ONE google result that >revealed connection between enable_ir=1 and livedrive midi. It would help if enable_ir would be renamed as enable_livedrive_ir_and_midi. Then people would not need extra documentation. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

18 years, 8 months

[linux-audio-user] Re: Unofficial announce: Allo

by Juhana Sadeharju

>From: james(a)dis-dot-dat.net >> >> http://www.funet.fi/~kouhia/allo20050819.tar.gz > >Very nice. Any hints on getting this into wav format? I'm sure >aplay/arecord would somehow manage it, but so far all I managed was >making it into hissing. It is 16-bit stereo CD audio: cat WORDS | allo | aplay -f cd It is also the same what is inside WAV format. Audacity's raw import quesses the format correctly. I could add a proper libsndfile support soon. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

18 years, 8 months

[linux-audio-user] Bounces from gmail ('suspicious headers')

by Dave Price

Greetings, I love this list and find it informative and helpful. One 'gripe': I can't seem to post to the list from google's mail system. Even though this is my subscribed address, I always get a bounce message when I post a message or reply using Google's Web client. The bounce indicates that my post has 'suspicious headers'. I am able to post with the from/reply address to set to my google address out of a mutt client - however this is a pain, and not always practical (when I am away from my desktop). I realize that the moderator(s) of this list are volunteers and probably very busy with their real life, but not of the 'waiting for moderation' messages ever seem to make it thru, so even if I were willing to be patient, I would not likely get a reply to my post ever, if I did not use a different client. Is there anything that can be done to adjust the automatic bounces to accept google mail's headers? I use the same address and client to work with several other lists with no problems. aloha (and thanks for the great list), dave

18 years, 8 months

[linux-audio-user] ALSA mixer issues - update

by Dave Price

Lee, Getting close. Setting the mpd device as you suggested makes the PCM slider work correctly while mpd is playing. mpd has a volume control which I can access using phpmp, but this does not seem to change the volume level, as it does on systems with different hardware. There are some mixer settings I can play with in mpd that may make this work, but any clues as to where to start reading besides the man page would be appreciated. In xmms, when I configure the plugin to use the defualt device, the slider on the xmms display and the alsamixer PCM slider move together, but the volume does not change. There is a 'software volume control' checkbox in this configuration which does work. - Hide quoted text - On 9/1/05, Lee Revell <rlrevell(a)joe-job.com > wrote: On Thu, 2005-09-01 at 15:37 -0600, Dave Price wrote: > ao_driver_options "dev=hw:0,0;buf_size=4096" > That's wrong, the device should be "default" not "hw:0,0". Lee -- aloha, dave

18 years, 8 months

[linux-audio-user] Re: Resampling Audio Libraries & Sinc Resamplers

by davidrclark＠earthlink.net

Part 2 of (maybe) 3 For those with lack of familiarity with Fourier analysis and synthesis, here is a concrete example to demonstrate potentially serious problems with sinc resamplers in doing bulk conversions at constant rates. These problems are real and could easily result in audible artifacts --- something that I assume is of importance to Linux audio users --- and especially with further processing. ------------------------ File 123163main_cas-skr1-112203.wav is the NASA file recently mentioned on LAU --- a public-domain, taxpayer-supported WAV file sampled at 5000 samples per second. This file was chosen arbitrarily --- just happened to resample it before reverbing and posting for interested LAU'ers a while ago, so decided to use it for a comparison for Steve Harris. Two resamplers: 1) sinc resampler: $ sndfile-resample -to 44100 -c 0 123163main_cas-skr1-112203.wav \ saturn_sndfile-resample.wav 2) FFT with large windows: Sampster in Mixster (stuff I wrote myself) Comparison was every 50th sample in the original file with every 441st sample in the other two (should match exactly every 0.01 seconds) for the first 9.5 seconds of the files. 9.5 seconds was chosen rather arbitrarily --- nothing special about it. Ideally these particular samples should match exactly. Any error indicates corruption of the original data at the exact locations where the original samples were taken. The last two columns show you the difference between what is expected at these matching points and what was actually obtained after resampling. Note that the values in the last column are significantly greater than those in the next-to-last column. Match# Original FFT sndfile # FFT sndfile (diffs) 1: 0.00000 -2.00000 19.0000 1: -2 19 2: 386.000 384.000 437.000 2: -2 51 3: -181.000 -183.000 -178.000 3: -2 3 4: -500.000 -502.000 -538.000 4: -2 -38 5: -1065.00 -1067.00 -1068.00 5: -2 -3 6: -54.0000 -56.0000 -28.0000 6: -2 26 7: -120.000 -122.000 -55.0000 7: -2 65 8: -348.000 -350.000 -344.000 8: -2 4 9: 827.000 825.000 805.000 9: -2 -22 <snip> 344: -67.0000 -71.0000 100.000 344: -4 167 345: -378.000 -382.000 -275.000 345: -4 103 346: -37.0000 -41.0000 -101.000 346: -4 -64 347: -209.000 -213.000 -19.0000 347: -4 190 348: 269.000 265.000 86.0000 348: -4 -183 349: 62.0000 58.0000 27.0000 349: -4 -35 350: 427.000 423.000 446.000 350: -4 19 351: 154.000 150.000 -47.0000 351: -4 -201 352: 619.000 615.000 52.0000 352: -4 -567 353: -202.000 -206.000 111.000 353: -4 313 354: -366.000 -370.000 205.000 354: -4 571 <<< OUCH! Hope this doesn't get expanded. Over 100x larger error. 355: -146.000 -150.000 8.00000 355: -4 154 356: 549.000 545.000 558.000 356: -4 9 357: 279.000 275.000 -34.0000 357: -4 -313 358: -110.000 -114.000 -12.0000 358: -4 98 359: -184.000 -188.000 199.000 359: -4 383 360: -215.000 -219.000 -417.000 360: -4 -202 361: 244.000 240.000 74.0000 361: -4 -170 362: -474.000 -478.000 -152.000 362: -4 322 363: 188.000 184.000 562.000 363: -4 374 <snip> 938: -1448.00 -1449.00 -1468.00 938: -1 -20 939: -1203.00 -1204.00 -1161.00 939: -1 42 940: 3210.00 3209.00 3111.00 940: -1 -99 <<< about 100x larger error at 10% full scale 941: 5767.00 5766.00 5838.00 941: -1 71 942: -656.000 -657.000 -628.000 942: -1 28 943: -5165.00 -5166.00 -5163.00 943: -1 2 944: 1547.00 1546.00 1584.00 944: -1 37 945: 4410.00 4409.00 4445.00 945: -1 35 946: 1912.00 1911.00 1881.00 946: -1 -31 947: 5947.00 5946.00 5829.00 947: -1 -118 <<< Over 100x larger error at 18% full scale. 948: 5923.00 5922.00 5902.00 948: -1 -21 949: 3462.00 3461.00 3494.00 949: -1 32 What this shows is that at every 0.01 seconds, where the original file and the resampled file should have the same exact value (if the original data were preserved), large errors occur for sndfile- resample. ------------------------ resample-1.7 was even worse with a phase shift on top of this type of inaccuracy, coupled with rather serious spectral leakage beyond 2.5 kHz which was the original band limit (or might as well be assumed to have been). Upon examining the waveforms, I could see that resample-1.7 was doing an excellent job of tracing out the original waveform by drawing pretty much straight lines between points. Although visually reassuring, this actually adds spectral components that were not in the original. So it depends on what you want. This resampling probably won't sound like the original, but does look good in an editor. ------------------------ Also of interest is that the very latest version of sndfile-resample gives slightly different results than an earlier version for the locations which should match (the versions are for libsamplerate): Match# v 0.0.15 v 0.1.2 1: 19.0000 18.0000 2: 437.0000 436.0000 3: -178.0000 -179.0000 4: -538.0000 -539.0000 5: -1068.0000 -1068.0000 6: -28.0000 -28.0000 7: -55.0000 -55.0000 8: -344.0000 -344.0000 9: 805.0000 805.0000 10: -81.0000 -82.0000 11: 482.0000 482.0000 12: 78.0000 77.0000 13: 227.0000 227.0000 14: 501.0000 500.0000 15: 13.0000 12.0000 <snip> So the *amount* of corruption of the original data at locations which should match varies with version! Fortunately (or perhaps unfortunately depending upon your point of view) this latest version never varies more than 2 from the earlier version, so the "latest and greatest" is just as bad. The errors in the table above would be altered by 2 or less, which is insignificant.

18 years, 8 months

[linux-audio-user] lau illuminati

by Dan Easley

I've discovered the international symbol for the LAU Illuminati on, of all places, my very own web page. And I'm just a lurker! How dastardly! At any rate, here it is. http://towndowner.com/lau-illuminati.jpg well, I thought it was funny. I'm a terrible photoshopper, and I've been reading way too much fark/somethingawful. this mailing list and its participants are fantastic - i read and learn from you guys every day - thank you. keep illuminatizing! -- dan easley (dan(a)burntpossum.com) prabob

18 years, 8 months

[linux-audio-user] ALSA mixer issues - update

by Dave Price

Update: I rebuilt ALSA to 1.09b and rebooted. I now have mixer 'sliders' in alsamixer for master and PCM, however. Neither slider actually has any impact on the sound lever in my speakers. Muting the PCM channel actually stops the sound. Muting the MASTER channel has no effect at all. I am testing with mpd (music player daemon) set to output as follows: o_driver "alsa09" ao_driver_options "dev=hw:0,0;buf_size=4096" I have also tested with XMMS and the Alsa Plugin v1.2.10 [libALSA.so] with the same results. Any hints / clues would be appreciated. aloha, dave Here are the proc entries for my system: /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.9b. Compiled on Sep 1 2005 for kernel 2.6.11. /proc/asound/cards 0 [SI7012 ]: ICH - SiS SI7012 SiS SI7012 with CMI9761 at 0xdc00, irq 10 /proc/asound/devices 25: [0- 1]: digital audio capture 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 0: [0- 0]: ctl 33: : timer /proc/asound/pcm 00-00: Intel ICH : SiS SI7012 : playback 1 : capture 1 00-01: Intel ICH - MIC ADC : SiS SI7012 - MIC ADC : capture 1 /proc/asound/oss/devices 12: [0-12]: digital audio 3: [0- 3]: digital audio 0: [0- 0]: mixer

18 years, 8 months

[linux-audio-user] SIS (SI7012) sound but limited mixer

by Dave Price

Hi, Music is working find and surviving reboots and restarts of KDE - now I am just wanting to fine tune things. alsamixer sees my (onboard) sound hardware as: Card: SiS SI7012 Chip: C-Media Electronics CMI9761 My biggest problem is that the mixer does not offer any range of control over the volume - I can mute it, but not adjust it on the card. The Master and PCM 'sliders' in alsamixer just say '00' and no slide is available. Here is my lsmod (at least the sound part): sis900 17796 0 snd_intel8x0 28992 4 snd_ac97_codec 74232 1 snd_intel8x0 snd_pcm_oss 48416 1 snd_mixer_oss 17792 3 snd_pcm_oss snd_pcm 83720 3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss snd_timer 22020 2 snd_seq,snd_pcm snd 48484 11 snd_seq_oss,snd_seq,snd_seq_device,snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 7776 4 snd snd_page_alloc 7812 2 snd_intel8x0,snd_pcm Any ideas what I should adjust to get a mixer/volume control to do anything meaningful? aloha, dave

18 years, 8 months

re:[linux-audio-user] Saving Alsa Mixer Settings

by Dave Price

Nigel, Yes, I am using KDE. I have now read the alsactl man page, and run (as sudo ) 'alsactl store' When I restart X/KDE (ctrl-alt-back) my sound cuts off as KDE initializes, indicating that the capture channel is unmuted. But if I then bring up a console and run 'alsactl restore' - even as a normal user, sound output returns. (I am running mpd at bootup, so it is pretty obvious when the sound stops / starts) As an aside, mpd rocks for playing mp3 files. I usually 'control' it with phpmp from my web browser. Thanks for the reply. aloha, dave PS: I have to send to the list from mutt, because google mail seems to put 'questionable headers' into my messages, which gets 'em suspended. So if the threading is funny, that's why. On 8/31/05, nigel henry <cave.dnb(a)tiscali.fr> wrote: Hi Dave. Are you using KDE by any chance, as it's likely to change mixer settings after alsactl has restored your settings. Nigel.

18 years, 8 months

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Linux-audio-user September 2005