similar to: sound library

Dear All, In the last few days I started using the last version of R as I encountered a problem with R1.9 and the use of RWinEdt, however the problem shows with R2.01 as well. More precisely 1. I install R2.01 after removing old verions of R 2. I Install RWinEdt package versione RWinEdt_1.6-2 and SWinRegistry_0.3-2 and everything seems fine 3. I recall the RWinEdt library (library(RWinEdt)) and

Dear R helpers, I'm using R2.2 under windows XP professional on a Dell double processor workstation. I have a large (29Mb) workspace that I'm trying to load both, by double clicking and by direct load. The message I get is "Bad restore file magic number (file may be corrupted)...". I believe that this is a consequence of a huge spike in the electrical system that my voltage

I have three time series, x, y, and z, and I want to analyse the relations between them. However, they have vastly different resolutions. I am writing to ask for advice on how to handle this situation in R. x is a stimulus, and y and z are responses. x is a rectangular pulse 4 sec long. Its onset and offset are known with sub-millisecond precision. The onset varies irregularly -- it doesn't

Hi, Has anyone thought about using the multichannel format as a way of distributing "musician minus one" files? Eh? I hear you ask. Ok. Basically these are multitrack files but you can mute the solo track so you, the musician, can practice against the other "players". You can buy CD's of this nature, but of course, the solo instrument is on one channel and the others on

Hi list! I am currently investigating the possibility of allowing musicians to collaborate over the internet -- Musician A -------> internet --------> Musician B This diagram obscures quite a lot of detail. In reality, the datapath would look more like this: Musician A ADC Compression algorithm UTP (or perhaps RTP?) IP ADSL the Internet ....<snip>..... Clearly, this link would

The following is my R script which I am struggling with to assess ICESat data..perhaps it is the ID_min or ID_max that is wrong? I don't know, any help would be greatly appreciated :( # OPTIONS - CHANGE THESE VARIABLES IF NEEDED\par ######################################################################\par \par icesatfile <-

Hello, I am a newbie when it comes to Dovecot, and in following this ( http://workaround.org/articles/ispmail-etch/ ) document in setting these things up, I have run into an error that I am not quite certain what to do... I am running Ubuntu Gutsy Gibbon, and using 1.0.5 for Dovecot. The message I am getting when I try to 'restart' Dovecot (using '/etc/init.d/dovecot restart') is:

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single

Hi, I have been using speex version 1.0.5 on a text-to-speech program. Recently I upgraded to version 1.2beta1 and noticed that the waveform the I got after encoding and decoding on the new versions (beta1,beta2) is much more different than the original than on version 1.0.5. I also ran a PESQ comparison test on 700 voice samples and got better results in the older version (I used quality 9, and

haven't seen this one mentioned; here's a cool musician that's put some of his work online in ogg format. http://www.alphaconspiracy.com/ (follow the http://aux.planetmath.org/freeculture/ac/ link from the 26 aug news item and click on "cipher") -- slothradio. synthpop / 80's / good stuff http://www.slothradio.com/ <p><p>--- >8 ---- List archives:

Hi, Having worked on a 4 channel encode/decode system, which works just fine (using 4 channel audio on SGI systems). I now would like to think a little further about how 'n' channel encodes may be used for network collaboration in music. Let's imagine a 16 channel codec, and let's say we are working in a simple stereo in/out environment (i.e. like most consumer devices out

I've just posted a demo made with the rgl package to Youtube, visible here: http://www.youtube.com/watch?v=prdZWQD7L5c For future reference, here are the steps I used: 1. Design a shape to be displayed, and then play with the animation functions to make it change over time. Use play3d to do it live in R, movie3d to write the individual frames of the movie to .png files. 2. Use the

I''m trying to fully understand the subtleties of using alias_method_chain to refine method behavior versus redefining a method and calling super. I have a simple data model (a music collection) that I often use to evaluate web frameworks, and I''m trying to get it working on Rails 1.2.0RC1. Specifically, I have three classes which subclass a common Artist class: - Band

On Tuesday 18. September 2007 14:32:45 Ian Malone wrote: > On 11/09/2007, Ian Malone <ibmalone@gmail.com> wrote: > > On 11/09/2007, Daniel Aleksandersen <aleksandersen+xiphlists@runbox.com> wrote: > > > On Tuesday 11. September 2007 01:34:35 Ian Malone wrote: > > > > I'd be interested which ones. DC is a bit nebulous, but that gives > > >

Hi, I am not a developer, but a musician. Can Opus be used in combination with any existing video-chat app to provide much better quality for the musical instrument sound transmission? Many thanks, Rustem -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170713/484472c2/attachment-0001.html>

On Aug 7, 2009, at 21:48, Didier Dambrin wrote: > FLAC doesn't preserve every chunk? I thought it did. I only gave a > quick try > but it seemed to have preserved even the most obscure chunks. > Let me check: it even seems to preserve "MIDI note associated to > marker", > which is a very unknown metadata used by SoundForge (& even defined > in a >

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there

Dear all, I've converted a wav file to flac but during the process three wrnings were casted. One of them I recall, it said that a BEXT chunk is unrecognized and thus ignored. Wouldn't it be posible to keep the unrecognized information as is when decoding the file? In general the extra chunks are located before or after the audio content, and represents a tiny part of the whole

I am specifically using Celt Wideband (48kHz) over WiFi multicast that naturally leads to lost packets and am trying to minimize the impact to the audio. I implemented PLC but the audio it produces is robotic. Have I implemented PLC correctly? Checking the waveform it is using the previous received waveform to fill in a missing packet but not the full waveform so it has to repeat. Basically,

Hello everyone! Sorry, if the whole task is silly, I'm new to this. I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I have a simple German isdn line and I have a microphone, headphones and a running JACKd (JACK Aduio Connection Kit). The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card,