similar to: codec at very low bit rate (2 kbps)

I'm sorry, but I've found in the paper "Speex: A Free Codec For Free Speech" at page 5 the phrase "If more than one channel is used, then an instance is required for each channel (Speex is not stateless)". What does it mean? Could it be that is the problem that I've met? Thanks Andrea ---------- Original Message ---------------------------------- From:

Thanks for the answer. So data are stored in little-endian format on the WAVE file. But have I to encode in big endian format or little endian is ok?! Thanks ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Tue, 23 May 2006 12:15:05 -0700 >You are probably reading little-endian samples on a big-endian platform, and

Yes but I need to encode at 2Kbps.. are there other free codecs expecially projected for very-low but rate? ---------- Original Message ---------------------------------- From: Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> Date: Tue, 30 May 2006 00:51:25 +1000 >Le lundi 29 mai 2006 ? 16:45 +0200, Leonardi Andrea a ?crit : >> Hi! >> Is it possible to encode at a rate of

Hi to everyone. I'm using speex in an application at the minimum bit rate (2.15 kbps), setting the encoder quality at 0, but obviously the quality is terrible. My question is: there are some signal processing that I can do to improve the quality of the signal at very low bit rate? Or can you tell me other codec that works well at that rate? Thank you.

Hallo, yesterday I was puzzled when I discovered that I probabliy miss something in the interepretation of intercept in two-way lm models. I thought that the intercept, using the default contr.treatment contrasts, represents the mean of the group of observations having zero in all column of the model.matrix. It turns out not to be case To be more more clear I am attaching a short example:

Hallo, I have a question that is more statistic related than about nlme and R functioning. I want to fit a complicated nonlinear model with nlme with several different measures of transpiration taken on each of the 220 trees grouped in 8 families. The unknown parameters of the model are three + their variances (and covariances). I want to estimate the variances among families of the

Hi Jean-Marc, Thanks. I've got the code for sampleenc and sampledec. But I didn't know how to verify whether the decoded file is correct. So I tried to analyse the decoded output and compile with my input. Test case: PCM samples, 8 bit, 8000 samples per sec. When I pulled all my input to 0s, after encode and decode, the output fluctuates around -0.12 to 0.2 (in float), so when finally

Dear R-Help Members, I am trying to read and analyse a set of 100 csv files. I need work only on some columns in each of those, so I decided to use a for loop, isolate the column in each file to work on, but then an error mesage appear: "undefined columns selected" Here is my code: setwd("F:/Data/") a<-list.files() for (x in a) { ? u<-read.csv(x, header=T,

Hi Jean-Marc, I finally make it work! Thanks for telling me that the algo will not work for 8-bit PCM. In fact I've tried using 16 bit PCM last time, but apparently there's a bug. The bug lies with the RecBuffer. The RecBuffer must be set to 320bytes instead of 160 bytes since every sample is now 16 bit. The compression is amazing, from 320 bytes to 38bytes! Thanks a million, Carine

I have this warning when I mount MS Windows shares, tunneling SMB over SSH in order to create a VPN-like shares management. Here the steps: # ssh -2 -N -L 445:shares_server:445 user@shares_server And after creating the port forwarding: # smbmount //share_server_netbios_name/share_directory /mnt/Samba -o username=user%pass,ip=127.0.0.1 On the shares_server runs a MS Windows XP Professional,

hi all i need to setup a live web radio using icecast, currently ruuning with mpd and a list of files. I need also to mix voice and music how can I Do this? Have you some links to study? thanks __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi

Hi all i would like to know if there's a software to create a dj station taking the input from mic and stream with icecast, also i need to know if is possibe to mix mic input and ogg file and stream it togheter with icecast thanks __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i

Hi all i would like to use icecast to get the audio stream and send it to some mobile user using a streming server as Helix ( encoder or producer). do you think is possible to encode the audio stream for example in AMR? Z __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e

yes ! but the problem is that it's not easy to use onother player instead of real of real or wmp thank you ----- Messaggio originale ----- Da: Mark Trimble <trimblma@mscd.edu> A: icecast@xiph.org Inviato: Mercoled? 15 novembre 2006, 14:59:46 Oggetto: RE: [Icecast] using some streaming server for mobile user I use Icecast2/Ices2 to broadcast two downmixed and resampled 16k mono ogg

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Hi, I'm still unable to get rid of the background noise though I've changed the PCM samples to 16 bit. I'm began to wonder if I can just take the PCM samples and encode them. What my application tries to do is to record the voice from a microphone, encode it, send it via the socket to a peer, decode at the peer's side and play the voice. My application works fine without the

Dear Andrea. Your request seems vague. And also: be careful, even if some of the people reading the list understand italian you are expected to post your requests in english. In short: read the FAQ and at least "An introduction to R". But you seem to be lucky, on the CRAN you will find a document by Vito Ricci, in italian, about time series analysis:

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Hi, I've changed the m_waveFormatEx.wBitsPerSample for both record buffer and playSound buffer, changed my array from char to short and I'm still getting the same constant background noise. I notice that when I talk into the mike, I will get a slightly louder noise, can't hear any clear speech... Any idea what's wrong? Regards, Carine ----- Original Message ---- From: John Miles