similar to: S4 helper functions: regular or generic?

Hi there, Problem :: When one tries to change one or some of the columns of a data.frame, R makes a copy of the whole data.frame using the '*tmp*' mechanism (this does not happen for components of a list, tracemem( ) on R-2.6.2 says so). Suggested solution :: Store the columns of the data.frame as a list inside of an environment slot of an S4 class, and define the '[',

Hi, Suppose I have a matrix of size 256x14x32 in MATLAB. I want to import that into R. I used readMat and then do.call. But the variable is stored as an array as its done in R. However, I want to define a variable W=array(0,c(256,14,32)) in R and read the multidimensional matlab variable into W. I am not able to do this in R. Please, could you help ? Thanks Gopi [[alternative HTML version

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see silk listed in menuselect as a codec. But I also don't see an asterisk 11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk. Do we use the asterisk 10 codec_silk.so ? sean

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

Hi, I want to configure dovecot as gmail imap proxy. If any system in my network wants to connect to imap.gmail.com,993 it has to go through the dovecot proxy. Is it possible to setup such environment with dovecot? If yes, can you please help me to configure my dovecot server. Warm Regards Supratik

Hi, I am running xen-unstable , now i have a Full Virt DomU running RHEL 3 update 5 wanted to know is there a way to convert this vm to Paravirt . regards Satya _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

I would like to know whether there is a faster way to do the below operation (updating vec1). My objective is to update the elements of a vector (vec1), where a particular element i is dependent on the previous one. I need to do this on vectors that are 1 million or longer and need to repeat that process several hundred times. The for loop works but is slow. If there is a faster way, please let

The following is mainly a capture of parts of multiple off-line discussions within members of the Crossbow team (Gopi, Thiru, Roamer, May-Lin, Thirumailai, Nitin, KB, ...), I thought I''d open it up to other participants. Crossbow''s core scheduling involves switching a NIC (or individual Rx rings on the NIC) to polling mode. The receive interrupt will become not only rarer,

I am trying to compare the speed of my R code with that reported by the authors of a reference paper who used Matlab. The authors of the paper state "Total CPU time for the algorithm was 10 s for the ...on a modest mobile 2.2 GHz Pentium processor running MATLAB v.6". I am using proc.time (similar to the example shown in ?proc.time) to determine the speed of my R code. Could someone

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work?

Another crash with a packet: $10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 324, offset = 64, src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318, uint32 = 2156475160, pad = "\030\063\211\200\000\000\000"}, delivery = { tv_sec =

Hi Gabriel, I m trying to author a test suite for video codecs. Can you guide me how to go about it? I need some ideas to get started with. Thanks Jina -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20050906/3c71295d/attachment.htm

When I activate the equalizer in WinAmp, QCD, and Sonic Foundry SIREN Jukebox, I get a funny "washed out" sound to it. Vorbis sounds fine without it, but I miss the extra bass and what-not. I was wondering why this happens? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

> A Siren codec is not currently available and the one for 12 will not > work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why).

I''ve been thinking about hardware that has multiple transmit rings ("tx resources"). We really should have a way to expose this up to the stack. And ideally, the stack should guarantee that a given flow will always be sent down using the same hardware tx resource. I''ve heard that crossbow will deliver this, but I can''t find evidence of it in the crossbow

Anyone have a status update on when Sonic Foundry plans on supporting RC2 ogg files? A while back on this list someone said they had a good contact at SF and was supposed to chekc up on it. I have contacted Ahead Software about including Vorbis support with Nero and they asked that a developer contact them about including it. Any chance a developer can contact them about support for Vorbis

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

Hi, On Fri, Oct 30, 2009 at 8:41 AM, Gopi Desaboyina <gopidesaboyina at yahoo.com> wrote: > I think your system might be over heating. I observed this kind of behaviour in laptops when they get overheated. check if FAN is working or not. How frequent it gets rebooted. you could boot from opensolaris LiveCD and keep it for a day like that. if it reboots that means there could be h/w