Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

3 participants
7528 discussions

[linux-audio-dev] Re:%20[linux-audio-dev]%20Echo%20Darla/Gina/Layla/...%20on%20Linux&replyto=200212060226.gB62QY48020703_AT_mail.space2u.com

by Shred Master

HELL YEAH!! - Shred. <a href="http://www.shred.ath.cx/">http://www.shred.ath.cx/</a> <a href="http://www.mp3.com.au/obsidian">http://www.mp3.com.au/obsidian - Obsidian - Original Hard Rock at its BEST!</a>

22 years, 5 months

[linux-audio-dev] Re:%20[linux-audio-dev]%20Echo%20Darla/Gina/Layla/...%20on%20Linux&replyto=200212060226.gB62QY48020703_AT_mail.space2u.com

by Shred Master

22 years, 5 months

[linux-audio-dev] gmorgan-0.01

by holborn

Hi! First Release. gmorgan is a .. Rhythm Station, an organ with auto-accompaniment. Uses MIDI and the ALSA sequencer for play the rhythm patterns. Styles, patterns and sounds, the mixer settings, can be edited and saved. Tested on Gentoo, debian PIII 933 and PII 300 REQUERIMENTS -------------------------- 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, 5 months

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

by Juhana Sadeharju

Could anyone check the shmserver.tar.gz, compile it, and test if you have the same problem? Additional instructions below. I have though about why the shared memory freezes. It could be that Linux copies the shared memory pages for each client: one shared memory segment, multiple copies. Would locking the memory help? Maybe, maybe not. I know there are high performance applications using the shared memory well (at least their authors may think so). But so does my own alsashmrec works well too (so I thought). If there indeed is a severe (a minor?) problem in the kernel, then in all these application it may cause very short freezes (until read, write, sleep, suspend, etc. call releases it). A jitter in updating lock-free FIFOs on shared memory. If you don't get it or otherwise cannot find out what is going on, then where I should post this? Is there a list where the kernel and shared memory experts are reading? And hey, why shmget did not work? See shmalloc_named_get(). Best regards, Juhana Additional instructions and the previous mail. 1. Compile it without sleep(1) [ as in code below ] and without fprintf with ggg's [ one could change that to "g" so that screen does not go too fast ]. Then compile and test with sleep, and then test with fprintf with "g". 2. First run shmserver in a terminal. Copy the mid value (second value printed). 3. Then run shmclient in another terminal. Paste the mid value: % shmclient <midvalue> >From: Juhana Sadeharju <kouhia(a)nic.funet.fi> >To: linux-audio-dev(a)music.columbia.edu >Subject: >Date: Fri, 4 Jul 2003 16:46:25 +0300 >Reply-To: linux-audio-dev(a)music.columbia.edu > >>From: Ralfs Kurmis <kurmisk(a)inbox.lv> >> >>Try out this example >>Split it in seperate files >>( Needs x ) > >Hello. Thanks for the example, but I see some problems there: >if the second process does not find the segment given by the key, >you example makes two distinct segments. That is what happen with >me. Because I don't have IPC_CREAT in the second process, my >program simply fails instead of creating second segment. > >I got it working otherway, but there are severe problems. >In client, I simply skipped the shmget() and queried >immediately the segment with shmat() with the mid value >printed by the server. > >The example mailed here used shmget() with IPC_CREAT. >When I used IPC_CREAT for both server and client, as I expected, >I got two separate shared memories. In fact, as I make the shared >memory in shmserver which is run first, the shmclient should not use >IPC_CREAT at all. > >It works, but while the server seems to fill the shared memory >with increasing integer numbers, the client behaves strangely. >I have this code in shmclient now: > > k = -1; > for (;;) { > if (k != nums[1]) { > k = nums[1]; > fprintf(stderr,"%i\n",k); > } > // sleep(1); > // fprintf(stderr,"ggggg\n"); > } > >What should it do? It should ideally print the increasing numbers: >5435, 5436, 5437, etc. With sleep(1) it prints a new value once per >second. However, without sleep(1), it prints only one number and then >does not print anything anymore. It looks like Linux does not update >the shared memory, why? > >When the "ggggg" is printed (without sleep), the shmclient prints only >one number and repeatedly the "ggggg". Why the shared memory is not >updated in this? I remember I had a similar problem with old XWave >software at 1998 with much earlier kernel version (now I have 2.4.18 of >RedHat 7.3). > >This looks serious problem. It may be that nobody has noticed >it because either one uses sleep() or read()/write() in an >audio system. That is, your software may work, but the problem >may degrage the performance (as it certainly did freeze the >printing in my shmclient). Perhaps the problem may cause an audio >engine never work as fully as possible. > >If you get the shmclient work while the sleep(1) is commented out, >please let me know :-) > >http://www.funet.fi/~kouhia/shmserver.tar.gz

22 years, 5 months

[linux-audio-dev] [ANN] VLevel 0.5

by Tom Felker

Hello all, I've been working on VLevel, a LADSPA plugin to keep me from having to fiddle with the volume, and it's now in a useful state, so I'm looking for some feedback. Basically, VLevel keeps track of the peak amplitudes, and adjusts the volume smoothly to make the quiet parts louder. Since it looks ahead a few seconds, the gain change is always smooth. <http://vlevel.sourceforge.net> VLevel is written in C++. I have two questions. First, why do most other plugins allocate and free copies of their strings and structures, instead of just passing the literal (as I do)? The declarations in ladspa.h don't allow the host to modify what the pointers reference. Second, I keep a buffer of length n in my code, so the first n seconds of data I return is useless, and after the audio is sent, I need n more seconds of input before all the audio is returned. Is there any way of informing the host about this? In the future I plan to make some performance improvements, and perhaps a nice cross-platform GUI for applying VLevel to files. I may also try to get XMMS-LADSPA to save its state, which would be very useful to me. I suppose VLevel could use RMS or a psychoacoustic model to estimate volume, but that would make it very complex, and more difficult to avoid clipping. Despite that, it serves my purpose, to play classical music on the road, quite well. Have fun, -- Tom Felker <tcfelker(a)mtco.com>

22 years, 5 months

[linux-audio-dev] It's been more than a year... Have things improved?

by Juan Linietsky

Hi everyone... I guess it's been more than a year since the last time we discussed such issues here. I am sure that everyone here, aswell as myself, works very hard to mantain and improve their respective apps. Because of it, the intention of this list post is to try to inform myself, aswell as possibly other developers about the status of many things that affect the way we develop our apps under linux. As many of you might remember, there were many fundamental issues regarding the apis and toolkits we use, I will try to enumerate and explain each one of them as best as I can. Please I ask everyone who is more up to date with the status of each to answer, comment, or even add more items. 1- GUI programming, and interface/audio syncronization. As well as I can remember, a great problem of many developers is how to syncronize the interface with the audio thread. Usually, we have the interface running at normal priority and the audio thread running at high priority (SCHED_FIFO) to ensure that it wont get preempted while mixing, specially when working with low latency. For many operations we do (if not most) we can resort to shared memory to do changes, as long as they are not destructive. But when we want to lock, it is most certainly that we will suffer from a priority inversion scenario. Althought POSIX supports functionality to avoid such scenarios from happening (Priority ceiling/inheritance), there are no plans to include support for such anytime soon in Linux (at least for 2.6, from what Andrew Morton told me). Althoght some projects exist, it will not likely to become mainstream for a couple of years (well, low latency patches are not mainstream either, with good reason). I came to find out that the prefered method is to transfer data through a FIFO (due to the userspace lock free nature), althought that can be very annoying for very complex interfaces. What are your experiences on this subject? Is it accepted to lock in cases where a destructive operation is being performed? (granted that if you are doing a mixdown you will not be supposed to be doing that) >From my own perspective, I've seen even commercial HARDWARE to lose the audio, or even kill voices when you do a destructive operation, but I dont know what users are supposed to expect. One thing I have to say about this also, is JACKit (and apps written for it ) low tolerance for xruns. I found many apps (or even JACKit itself) would crash or exit when such happens, I understand xruns are bad, but I dont see how they could be problem if you are "editing" (NOT recording/performing) and some destructive operation needs to lock the audio thread for a relatively long time. 2-The role of low latency/Audio and MIDI timing. As much as we love working with low latency (And I personally like controlling softsynths from my roland keyboard). In many cases, if not most? It is not really necesary, and it can be counterproductive, since working in such mode eats a lot of CPU out of the programs. Low latency is ideal when performing LIVE input and you want to hear a processed output. Examples of this are input from a midi controller and output from a softsynth, or input thru a line (guitar for example) and processed output (effects). But Imagine that you dont really need to do that.. you could simply increase the audio buffering size to have latencies up to 20/25 milliseconds, while saving CPU, preventing xruns, and the latency is still perfetly acceptable for working in a sequencer, for example or doing audio mixing of pre-recorded tracks. Doing things this should also ease the pain to softsynth writers, as they wouldnt be FORCED to support low latency for their app to work properly. And despite the point of view of many people, many audio users and programmers dont care about low latency and/or dont need it. But such scenario, at least a year ago, was(is?) not possible under Linux, as softsynths (using ALSA and/orJACKit) have no way to syncronize audio and midi, unless running in low latency mode, where it no longer matters (audio update interval is so small that works as a relatively high resolution timer). Last time I checked, Alsa could not deliver useful timestamping information for this, and JACKit would also not deliver info on when did the audio callback happened. I know back then there were ideas floating aroudn on integrating MIDI capabilities to JACKit to override this problem and provide a more standarized framework. I dont see how should also MIDI sync/clocks help in this case, since it's basically meant for wires or "low latency" frameworks. 3-Host instruments. I remember some discussion on XAP a while ago, but having been to the page recently, I saw no progrerss at all. Is there still really a need on this? (besides the previous point) or is it that ALSA/JACKit do this better, besides prooviding interface abstraction? Also, I never had very clear what is the limitation regarding the implementation of the VST api under linux, granting that so many opensource plugins exist. Is it because of the api being propertary, or similar legal reasons? 4-Interface abstraction for plugins.: We all know how our lovely X11 does not r allow for a sane way of sharing the event loop between toolkits (might this be a good idea for a proposal?) So it is basically impossible to have more than a toolkit in a single process. Because of this, I guess it's impossible and unfair to decide on a toolkit to configure LADSPA plugins from a GUI. I remember Steve Harris proposed the use of (rdf was it?), and plugins may also provide hints, but I still think that such may not be enough if you want to do advanced features such an envelope editor, or visualizations for things such as filter responses, cycles of an oscilator, etc. Has anything happened in the latest months regarding to this issue? 5-Project framework/session management. After much discussion and a proposal, Bob Ham started implementing LADCCA. I think this is a vital component, and will grow even more important granting the complexity that an audio setup can lead to. Imagine if you are runniing a sequencer , many softsynths, effect processors and then a multitrack recorder, all interconnected via ALSA or JACKit.. saving the setup for working later on can be a lifesaver. What is it's state nowadays? And how many applications provide support for it? Well, those are basically my main concerns on the subject, I hope to not have sounded like moron, since that is not my intention at all. I am very happy with the progress of the apps, and It's great to see apps like swami, rosegarden or ardour become mature with the time. Well, cheers to everyone and lets keep working hard! Juan Linietsky

22 years, 5 months

[linux-audio-dev] ableton live

by Adrian Gschwend

Hi all, Yesterday I visited a demo event of ableton live in Switzerland. I've read quite a lot about this thingy in magazines but I never tried it myself (I don't use windows since 1994 anymore). But man I was really impressed. This is by far the most intuitive sequencer for all kind of music I've ever seen. The concept is a bit hard to describe, if you know trackers from the good old days imagine that mixed with a real-time timestrecher for all your samples, a harddisk recording tool and many nice enhancements like effects, crossfader (dj mixtable like)... The best thing is if you download a demo at http://www.ableton.de/ ("Products->live->demo download") and give it a try, it should also work inside VirtualPC, VMWare or whatsover. The thingy is quite fast, even on old machines. anyway, after trying to find the most intuitive sequencer interface for years I think I've found that yesterday, too bad it was not my idea ;) So I'm tempted to start a project to write something like that as open source. The most important part in it is definitely the timestreching (they call it "elastic" audio...). But as far as I know timestreching algorithms are 1. not easy to implement and 2. not open source if they sound good :) Because I'm an absolute newbie with timestretching I request for comments. Can anyone point me to some papers or reference implementations of (realtime) timestretching algorithms? It won't be needed in a first stage of the application but in a long term it is a must. Also, if someone is working on something like a "live" for Linux let me know :) cu Adrian -- Adrian Gschwend @ netlabs.org ktk [a t] netlabs.org ------- Free Software for OS/2 and eCS http://www.netlabs.org

22 years, 5 months

[linux-audio-dev] linuxtag: any developers willing to q and a over irc ?

by Joern Nettingsmeier

hi everyone ! while i'm slowly getting in linuxtag mode, it occured to me it might be nice to direct booth visitors with very specific questions to the developers themselves... so if any of you are hanging out on #lad anyway, let me know when and where you can be reached (and which time zone you are in), it might be useful. in any case, it will be fun :) just a little idea... jörn -- All Members shall refrain in their international relations from the threat or use of force against the territorial integrity or political independence of any state, or in any other manner inconsistent with the Purposes of the United Nations. -- Charter of the United Nations, Article 2.4 Jörn Nettingsmeier Kurfürstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

22 years, 5 months

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

by Juhana Sadeharju

>From: Ralfs Kurmis <kurmisk(a)inbox.lv> > >Try out this example >Split it in seperate files >( Needs x ) Hello. Thanks for the example, but I see some problems there: if the second process does not find the segment given by the key, you example makes two distinct segments. That is what happen with me. Because I don't have IPC_CREAT in the second process, my program simply fails instead of creating second segment. I got it working otherway, but there are severe problems. In client, I simply skipped the shmget() and queried immediately the segment with shmat() with the mid value printed by the server. The example mailed here used shmget() with IPC_CREAT. When I used IPC_CREAT for both server and client, as I expected, I got two separate shared memories. In fact, as I make the shared memory in shmserver which is run first, the shmclient should not use IPC_CREAT at all. It works, but while the server seems to fill the shared memory with increasing integer numbers, the client behaves strangely. I have this code in shmclient now: k = -1; for (;;) { if (k != nums[1]) { k = nums[1]; fprintf(stderr,"%i\n",k); } // sleep(1); // fprintf(stderr,"ggggg\n"); } What should it do? It should ideally print the increasing numbers: 5435, 5436, 5437, etc. With sleep(1) it prints a new value once per second. However, without sleep(1), it prints only one number and then does not print anything anymore. It looks like Linux does not update the shared memory, why? When the "ggggg" is printed (without sleep), the shmclient prints only one number and repeatedly the "ggggg". Why the shared memory is not updated in this? I remember I had a similar problem with old XWave software at 1998 with much earlier kernel version (now I have 2.4.18 of RedHat 7.3). This looks serious problem. It may be that nobody has noticed it because either one uses sleep() or read()/write() in an audio system. That is, your software may work, but the problem may degrage the performance (as it certainly did freeze the printing in my shmclient). Perhaps the problem may cause an audio engine never work as fully as possible. If you get the shmclient work while the sleep(1) is commented out, please let me know :-) http://www.funet.fi/~kouhia/shmserver.tar.gz Best regards, Juhana

22 years, 5 months

[linux-audio-dev] 2nd LAD conference in Karlsruhe

by Matthias Nagorni

Hello, as both the participants of the 1st LAD conference and people at ZKM enjoyed the meeting, Frank and I asked for the possibility to hold a second meeting at ZKM next year. The answer was positive and therefore we can announce that the 2nd LAD conference is planned to take place April 29th - May 2nd 2004 at ZKM Karlsruhe. the option to have more room. In addition to the rooms we had for the last conference, we now have the option to also use a hall which is about double the size of the lecture hall we used for the last meeting. This hall is even more attractive since it is the recording studio of ZKM and can also serve as a concert hall. This time there is also the option to invite artists who actually do music with Linux software. Early registrations (email either me or Frank Neumann <Frank.Neumann_AT_st.com>) would help us to estimate the approximate scale of the event which can be even larger than last time. If you can do a talk or presentation please let us know the subject and estimated time you need for this. Depending on the number of talks we can decide whether we will have two parallel sessions. If the program of the event is fixed earlier than last time, this will help to advertise it in journals and among relevant companies. It might also help to find possible sponsors. Updates on this will follow from time to time. Matthias -- Dr. Matthias Nagorni SuSE Linux AG Deutschherrnstr. 15-19 phone: +49 911 74053375 D - 90429 Nuernberg fax : +49 911 74053483

22 years, 5 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev