Linux-audio-dev

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7501 discussions

Re: [linux-audio-dev] LL-patch for SuSE kernel?

by Stefan Nitschke

>yes, suse kernel (since 8.1) already includes most of the >necessary changes. some parts are missing but they are on >the rare code path, which has been not audited quite well, >anyway. Well i do have a LL patch for the original SusE 8.1 kernel that comes with the distribution, including the capability patch. The original SuSE 8.1 kernel does not work well without that patch on my machines. I also have a patch that adds cpufreq and a patch for BIOSes with broken ACPI. That patches are only tested on my Dell i8500, but for me they work perfect. I dont have patches for SuSE 8.2 ... i wait for an update until the first stable 2.6.x kernel based SuSE distro is out. - Stefan _________________________________________________________________ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail

21 years, 8 months

[linux-audio-dev] mixing algorithms...

by David E. Storey

Hey there... I have an incredibly naive question... How does one mix n channels of audio down to one channel? I've scoured the 'net as best I could and haven't really found anything very authoritative. Suppose I'm dealing with float point data between -1.0 and 1.0 and have n channels. I know physically, it's all a summation of the individual waves, but strictly, summation of multiple waves between -1.0 and 1.0 doesn't keep you between -1.0 and 1.0. Looking through sox, they tend to multiply by some gain value relative to the number of channels. (namely... an average) So is averaging considered a valid professional audio method of digital mixing? what if your source samples are all 16bit? Wouldn't you need to dither in a case like that? Doesn't averaging also imply truncation of your source signal? Am I missing something here? I've tried looking to see what jack and ardour do, but was lefting wondering WHERE to look. respectfully, d!

21 years, 8 months

[linux-audio-dev] M-Audio MobilePre/Sonica/Transit/Ozone testers wanted

by Clemens Ladisch

The M-Audio MobilePre/Sonica/Transit/Ozone devices need a firmware download before they can be used with Linux. I've released a first beta of the madfu-firmware package which tries to accomplish this. Please note that I do not have any of these devices, and that the loader is not tested at all. That's why I'm searching for volunteers. :-) To download the beta, go to http://sourceforge.net/project/showfiles.php?group_id=87777&release_id=1833… Please send any success/failure reports to usb-midi-fw-user(a)lists.sf.net Regards, Clemens

21 years, 9 months

[linux-audio-dev] midi controls API, is it possible?

by antoine rivoire

Hi all, I am working towards using a set of linux audio apps in a live context. With recent development of Jack and apps such as djEQ and such, things look promising. In this light, an idea came to mind. (Keep in mind that I am thinking aloud, and that I am no expert in audio or midi programming). How feasible would it be to implement and a Midi controller API to have a standard, graphical ( a la qJackconnect) application API that Synths and such could use to change midi controllers assignment on the fly or load existing presets? I don't think that the audio framework that has been worked on in linux had live use in mind, but I am convinced that it lends itself to it. In the same context, and this might already be possible, is it possible to have a script calling different apps with patches/presets for each one of those? I am thinking along those lines: during performance: Call script -> opens say Zynaddsubfx with a certain master, Freqtweak with a certain session, Hydrogen with a certain song etc -> call end of piece from script which exits all these apps -> call another script for next piece. Regards Ant- -- antoine rivoire <antoine.rivoire(a)ntlworld.com>

21 years, 9 months

Re: [linux-audio-dev] What's the best audio IO API on Linux

by hexe_2003＠directbox.com

-------- directBOX Reply --------------- From: harold_zhu(a)hotmail.com To : linux-audio-dev(a)music.columbia.edu Date: 15.09.2003 04:13:24 <p>Hi, Folks: <p>I am new here and I am also kind of new to Linux, so I have a basic question. I have worked on a multi-media project on Windows platform, and I used WaveIn and WaveOut functions as audio IO interface to capture and playback real-time audio. Now, I need to do a port to Linux.......so what's the best audio IO API on Linux? <p>In my search so far, I understand that both OSS and ALSA can do the job (real-time audio capture/play).......but what's the difference? Stability wise? Ease-of-use? Performance wise? Also any other candidate, in particular, cross-platform wrapper API? <p>Thanks Express yourself with MSN Messenger 6.0 -- download now! ---- Hi! Well, the main differences between the OSS-Layer and ALSA are: 1.) alsa is the new driver in linux kernel >= 2.6.X 2.) oss-layers need ioctl() - calls to manage/configure sound devices (and ioctl() calls are just possible for root-users !!!!!),so oss is really user-unfriendly if you want to programm a application 3.) alsa is easier to use for midi / pcm capturing/playback 4.) both are less documentated ( I can tell you that !!! :) ) , but alsa has a few basing programming tuts for C ( c++) 5.) alsa is basicly said a "wrapper", a front-end of the oss-arch and it is highly recommended to use alsa at all. I hope this helps, Sascha Retzki __________________________________________________ Verpassen Sie keine eBay-Auktion und bieten Sie bequem und schnell über das Telefon mit http://www.telefonbieten.de Ihre eMails auf dem Handy lesen - ohne Zeitverlust - 24h/Tag eMail, FAX, SMS, VoiceMail mit http://www.directbox.com

21 years, 9 months

Re: Re: [linux-audio-dev] more practical stuff about synth-programming -- jack seems to be less documentated

by hexe_2003＠directbox.com

-------- directBOX Reply --------------- From: paul(a)linuxaudiosystems.com To : linux-audio-dev(a)music.columbia.edu Date: 18.09.2003 15:00:29 >It allows you to get a basic DSP algorithm producing sound without you >worrying about how to configure the audio interface, and then later >lets that same output be (potentially) routed to other applications >for processing/manipulation/recording. Well, this sounds very interesting; but where can I find developers documentation about jack ? Like I said, the Homepage is not interesting for that part of information need... . And has anybody some interesting c-algorythm documentation on how to produce convolution ? I mean: oscilating basing wave -> mix it together with a lfo -> filter it with delay, for example -> ... greetz, Sascha Retzki __________________________________________________ Verpassen Sie keine eBay-Auktion und bieten Sie bequem und schnell über das Telefon mit http://www.telefonbieten.de Ihre eMails auf dem Handy lesen - ohne Zeitverlust - 24h/Tag eMail, FAX, SMS, VoiceMail mit http://www.directbox.com

21 years, 9 months

Re: [linux-audio-dev] lowlatency test at linuxdevices

by Benno Senoner

Paul Davis wrote: > I hope this is not true: > > "Embedded systems often need to poll hardware or do other tasks on a > fixed schedule. POSIX timers make it easy to arrange any task to get > scheduled periodically. The clock that the timer uses can be set to > tick at a rate a fine as one kilohertz, so that software engineers can > control the scheduling of tasks with precision." > The promise of the high-res timer patch was usec resolution, not msec. > This would be a great loss. Does anybody know any more? Yes unfortunately it is true what they say in the article. The current timer resolution is 1msec (HZ = 1024, so to be precise the resolution is (1/1024) sec). In short the story is as follows: Linus accepted the POSIX 1003.1b Section 14 (Clocks and Timers) API in kernel 2.6 but not yet it's implementation (patches available here http://high-res-timers.sourceforge.net/ ). This means that applications using the POSIX 1003.1b timer API can specify timing values nanosecond resolution but for now only msec resolution is provided. But when the Linus & co will let in the kernel the high-res timer implementation, those apps will instantly be able to achieve higher resolution without recompilation etc. Yes usec resolution would be handy for some audio apps but I for now I am happy of being able to achieve msec resolution in MIDI playback without resorting to the RTC device which cannot easily be shared. PS: in the article they talk about 4500usec worst case scheduling latency (= 4.5msec), seems a bit disappointing. I'm curious what they mean with worst case, which kind of test suites they used etc. 2.4 + some LL patches let you reliably work with sub 3msec BTW. Benno ------------------------------------------------- This mail sent through http://www.gardena.net

21 years, 9 months

[linux-audio-dev] [ANN] Lemux version 0.2

by Joost Yervante Damad

Lemux is a collection of (GPL) LADSPA instruments based on devices from the openMSX emulator and other sources (e.g. sidplay2). Changes against 0.1: - all instruments are working, currently: - SCCChannel (e.g. from Konami games) (a 32 byte 8bit looped samples instrument) - PSGChannel (the standard sound chip from MSX) (a 1/2 square wave with noise and AM) - MUSICChannel (the FM OPLL chip from MSX-MUSIC) (a 2 operator FM chip, with 15 standard instruments and 1 custom FM instrument that is fully configurable) - MUSICDrum (FM OPLL Drums) (5 standard drum sounds from OPLL) - SID (The full C64 audio chip) - instrument volumes are now normalized More info (and audio samples) can be found at the website: http://lumatec.be/joost/lemux/ Greetings, Joost Damad

21 years, 9 months

[linux-audio-dev] Streaming compressed audio from the network

by Juhana Sadeharju

Hello. If I understood correctly, several companies are wanting patent royalties for streaming compressed audio from the network. Take a look at mpg321 (GPL clone of mpg123): Usage: mpg123 [options] file(s) | URL(s) | - So, this playing from the network URLs is no no, or what? What other software are able to play a compressed audio from the network? Juhana

21 years, 9 months

[linux-audio-dev] next power of two

by Maarten de Boer

Hi, A colleague of mine found this very clever algoritm to calculate the next power of two for a given number. It comes from the Prophecy SDK for 3d game development http://www.twilight3d.com/modules.php?op=modload&name=Downloads&file=index Since this is a kind of thing often needed in audio processing, I wanted to share it with you. I certainly can not think of a faster (or more elegant) way of doing it. Maarten ////// /// Returns the closest power-of-two number greater or equal /// to n for the given (unsigned) integer n. /// Will return 0 when n = 0 and 1 when n = 1. ////// inline uint32_t nextPowerOfTwo(uint32_t n) { --n; n |= n >> 16; n |= n >> 8; n |= n >> 4; n |= n >> 2; n |= n >> 1; ++n; return n; }

21 years, 9 months

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Linux-audio-dev