similar to: Wine apps in realtime,possible?

Ubuntu 10.04. Wine 1.2 I'm running Native Instruments Kontakt 4 under Wine. It runs fine except for occasional stutters, so I wanted to try enabling Realtime in Jack, but I have no luck with this. (I have made recommended edits to the limits.conf file, though I've read that this is no longer necessary(?): @audio - rtprio 100, @audio - memlock 250000) When I start up Kontakt, I see

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I was wondering if anyone had any advice on an audio library that is better suited to be paired with Celt in terms of latency. I'm working on an application that I would like to have running on both windows/Linux systems. I started by using the OpenAL library but I have run into an issue when feeding OpenAL small mono sample sizes that

In winecfg is alsadrv selected but i cant see any asio audio device in a asio compatible aplication and no ports in jack too if i do: env WINEDEBUG=+asio wine '/home/xxx/.wine/drive_c/Program Files/FXpansion/Geist/Geist.exe' Code: p11-kit: couldn't load module: /usr/lib/i386-linux-gnu/pkcs11/gnome-keyring-pkcs11.so: /usr/lib/i386-linux-gnu/pkcs11/gnome-keyring-pkcs11.so: no se

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

Thank you. But it will cost you a long time to get the accurate play and capture frequencies. Does your program test two frequencies of the sound card each time? Because different sound cards have different frequency errors. And the resampling program is also time consuming because the target frequency is so close to the sampling frequency of the input signal, isn't it? I have tested program

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: comunip.gif Type: image/gif Size: 1663 bytes Desc: not available Url : http://lists.xiph.org/pipermail/speex-dev/attachments/20090811/ad615b2a/attachment.gif

Hello, I'm trying to set up winamp to stream a live event to an icecast2 server. I've got the icecast2 server set up, and to date have streamed playlists only via ices0 or ices2, now I'm wanting to stream live using winamp. Software wise, I'm needing to use free or open source software, the icecast2 server is a Linux machine, the Winamp will be on either a Win7 or a WinXP

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

Hi All, In order to deal with acoustic echo cancellation problems of most PCs which sound cards have different capture and play frequencies. I made a trial. At first, a 1000Hz sine wave is played for a long time via a speaker and its acoustic echo is recoreded. Seconds, get the frequency of the echo by a FFT analyser. So the difference between capture and play frequencies is obtained. Thirdly,

Ubuntu 10.4, Wine 1.1.42, Native Instruments Kontakt 4, Akai MPK88 I've installed Kontakt (sampled musical instruments). Sound is working OK with the OSS driver, but the program doesn't respond to the USB keyboard. :- ( The keyboard shows up in `winecfg` under "Audio", OSS, MIDI In/Out devices. I checked the MIDI options for the Kontakt program and it DOES show the Akai

use icecast+liveice and i need to get stream from linein . Icecast work perfectly and liveice was perfectly but i can get stream only if I plug lineout from my radio box to mic on my sound card . Is there any special config string in liveice for streaming from linein or maybe liveice can't do it my liveice.cfg # liveice configuration file SERVER localhost PORT 80 NAME My Radio Box GENRE

I actually forgot to mention that I'm using ultra-wideband mode, but seems like you understood that anyway. Is this true that Speex echo cancellation only performs well in narrowband mode !? I've been using 100 ms as the default tail length. I don't know what the ideal tail length would be. I have tried shorter and longer tails but it hasn't made any difference. Does

Does wine asio still work with the current version of WINE? I tried installing it per the instructions and it didn't want to make.

Ive installed Winamp 5621 in Wine 1.3.28 using these instructions http://appdb.winehq.org/objectManager.php?sClass=version&iId=2723 but receive the following errors when trying to run it. Code: sevy at sevy-desktop:~$ env WINEPREFIX=~/.wine-winamp wine 'C:\Program Files\winamp\winamp.exe' fixme:ntoskrnl:KeInitializeSpinLock stub: 0x548024 fixme:advapi:RegisterTraceGuidsW (0x100778a,

Ok, this is slightly offtopic, but relates to the quality of input for speex :) I'm working on echo cancellation by means of sampling the wave mix of the sound card as well as the microphone. I originally had two sound cards, which had some synchronization problems (now solved, more or less), but I have also discovered a much better solution using ASIO 2.0, which enables me to sample

From: BILL OATMAN I have seen several requests for windows specific example code. This is another. What I am looking for is a example of how to play a large ogg file in windows by using a multi-buffering technique. That is, reading and decoding a chunk of the ogg file and handing it off to windows (directsound maybe) meanwhile buffering up the next chunk to play etc. Assume the ogg file

To everyone on the list: do *NOT* attempt to do echo cancellation with signals sampled using different clocks. This will *NOT* work. Just a 0.1% difference between the two sampling rates (it's sometimes worse than that) means that the impulse response drifts by 8 samples every second. There's just no way to efficiently track this. Or at least no way that doesn't involve something 100x

Yesterday I tried to compile wine asio.I am running debian Squeeze and tried both the lenny and sid wine repos..but on compiling wine asio.it errors out and cant find ..now this was yesterday so i cant remember the exact name ..but it was something like lobc32 ..something like that..maybe logbc32..but anyway..I uninstalled wine and wine-dev..and installed debian version of wine..it 1.0.1 along

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear