Linux-audio-dev

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7516 discussions

Re: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

by Mickael S Vardo

I'm a professional audio engineer for 15 years, and DSD did convince me. I could compare the sound quality with TI ADC/DAC. Conventional PCM techniques are unable to reproduce high frequencies correctly. And the explanation is very simple. If you record a sound at 44.1 kss, you get a theorical frequency response of 0 - 22050 Hz. BUT to describe frequencies from 11050 to 22050 Hz, you can only play with a 4-sample long period. A 22050 Hz sine could be really accurate (one sample up, one sample down every 1/22050th second), and so is 11025. But intermediary frequencies introduces temporal aliasing, some metallic feeling due to temporal quantization. This is inherent to the very low sampling rate (96 kHz is just a bit better, but no miracle), which is unable to describe waveforms at high frequencies. Bad high frequencies temporal definition means bad transients. Anyone can notice it when he _actually_ hear and compare PCM and DSD. Stop speculative talking and try to get some real demo... -- mickael

22 years, 3 months

[linux-audio-dev] Redirecting the isdncard's output to the speakers and redirecting the soundinput from the mic to the isdncard.

by Jorge

Dear Sirs, dear Madams I've programmed an utility called kisdnmonitor, it tracks the calls you make/ receive. It's for KDE as the name suggests. There's a server as well, it's called isdnserver. You can get more infos on both on: http://www.elogix.ch/linux_en.html Now I would like to add telephony support to the programs, so that you can use the computer as a "deluxe" isdn phone. So far I've figured out that you can dial through the modem emulation /dev/ttyIx. You just properly configure the device and then with the following Hayes codes: ATS18=1 ATS14=4 ATD<the phone number> you can start a call. Everything is ok so far, the problem is that, obviously, no sound will automagically come out from the speakers and nothing that is brabbled through the mic will go to the other side. Basically, what I want to do, is to stream the sound from the isdncard to the speakers and from the mic to the isdncard. Vbox (a voice call programm) is able to save samples from the isdncard's input. I should mention as well that years ago, I had a program that came with the isdncard (for windoze) called RVScom that had the same functionality, and it worked (in walkie talkie mode, I guess that my isdncard isn't full duplex capable). Since 8 bits suffice, the processor load shouldn't be extreme. Has anyone an idea? Any help would be greatly appreciated. Thank you in advance Respectfully submitted George

22 years, 3 months

[linux-audio-dev] calling all Terratec EWS88MT / envy24 control users

by Fons Adriaensen

Hello list, I hope there are some other users of the Terratec EWS88MT card out there, because I seem to be in serious trouble - the card 'works' but not in a way that makes it useful. I've got a test program that outputs the same signal (1 kHz sine at -6 dB below peak level) to all eight channels. Measuring the output levels, I get wildly different values. Using envy24control to set the DAC output levels only seems to make things worse - there is no logical relation at all between the slider position and the actual level. For each of the four pairs of channels, the two level settings interact in a more or less random way, mostly by just switching the other channel off or on. After a few minutes twiddling, all sound stops until I reboot. Similar things seem to happen with the inputs, but I haven't had the time to investigate those. So I'm wondering what's wrong - is my new card not up to standards, or is there a problem with envy24control trying to control it ? Any help / hints / tips appreciated ! -- FA

22 years, 3 months

Re: [linux-audio-dev] Re: Shared memory

by Juhana Sadeharju

>From: ingo.oeser(a)informatik.tu-chemnitz.de (Ingo Oeser) > >For people like you gcc supports -fvolatile. > >Your code will be really slow, but you save typing 'volatile' where the >C-Compiler needs it. So, there is no "do not optimize away apperent volatile variables"? Volatile variables in shmclient are easily recognized -- at least I recognized them. It is the array variable which content is only read but never written. No such optimization flag? Can't believe that compilers are that dump. Juhana

22 years, 3 months

[linux-audio-dev] gmorgan-0.06

by holborn

Hi! gmorgan is a .. Rhythm Station, an organ with auto-accompaniment and a "small" Band in a Linux Box. Uses MIDI and the ALSA sequencer for play the rhythm patterns. Styles, patterns , sounds, and the mixer settings, can be edited and saved. Program is released GNU/GPL version 2. News on v0.06 --------------------- - Finally solved problem with CPU user system time. - Program runs with 2 threads less. - Added FIFO priority and memory lock. - Added Clear Bar in Pattern window edit for Bass,Acc1,Acc2 and Acc3. - Added small Help window in Pattern window edit. - Added Preset List Editor. - Added Note Drum Name Editor. - Added Program Change Slider. - Added "safe" Mode 2 for melodic accompaniments. - Added Patterns. - Minor bugs solved. REQUERIMENTS -------------------------- Fast Computer Linux ALSA Fltk Take a look at http://personal.telefonica.terra.es/web/soudfontcombi/ And please ... if you enjoy this prog and wants to share patterns, send me, and i will include in future versions, i have a large TODO, and i need some help. Josep

22 years, 3 months

[linux-audio-dev] wxWindows and Linux distros ?

by Dave Phillips

Greetings: I couldn't find the answer to this one on Google, so: Which Linux distributions include wxWindows in their default installations ? Best regards, == dp

22 years, 3 months

Re: [linux-audio-dev] Re: Shared memory

by John Lazzaro

> Paul Davis <paul(a)linuxaudiosystems.com> writes: > > you and the compiler need to be friends. you have to communicate, and > to share your strengths. neither of you are stupid, and the compiler > not a slave. Paul speaks with the voice of wisdom here, people, listen to him. Sfront is 100K+ lines of code in large part because it is trying to participate in two compiler-friendships: the SAOL programmer, and the C compiler ... ------------------------------------------------------------------------- John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro -------------------------------------------------------------------------

22 years, 3 months

[linux-audio-dev] kernel 2.6

by Tim Hockin

All, I haven't used kernel 2.5/2.6 for any audio stuff yet. I'm at the Linux Symposium this week - do we have any requests or gripes with 2.6 that I can relay to the core kernel guys? Audio is a workload they don't really test. Tim

22 years, 3 months

[linux-audio-dev] [ANN] QAMix-0.0.4

by Matthias Nagorni

Hello, Version 0.0.4 of the MIDI controllable QT Mixer for ALSA is available from http://www.suse.de/~mana/kalsatools.html There are now GUI descriptions for SB Live, RME Digi96 and a default description for AC97 compatible cards. The GUI has been improved and there are new XML tags. Check the README for more. Have fun ! Matthias -- Dr. Matthias Nagorni SuSE Linux AG Deutschherrnstr. 15-19 phone: +49 911 74053375 D - 90429 Nuernberg fax : +49 911 74053483

22 years, 3 months

[linux-audio-dev] Intro Text on filters

by Björn Pfeiffer

Hi all, here is the link to a IMO very fine introductory text on filters and the like. http://on1.zkm.de/zkm/Institute/Musik/ZKMusik/veroeffentlicht/PD_Filters They also have some other interesting texts, most written by Pierre Dutilleux, most in german: http://on1.zkm.de/zkm/Institute/Musik/ZKMusik/Veroeffentlichung/ <- bottom of page Most of you will already know the site, since they hosted the Linux Audio Developers Conference. -Björn Pfeiffer

22 years, 3 months

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Linux-audio-dev