Displaying 20 results from an estimated 27 matches for "finlayson".
2001 Nov 23
2
Rose diagrams in R?
...plot histograms of directional
data such as wind direction. I believe these are called rose diagrams. Is
there an R script for this? If not, can it be constructed in a function
calling primitive graphic calls (lines, circles, boxes or polygons)?
The stars function is not quite right.
--
David Finlayson
Geomorphogist and GIS Specialist
NearPRISM - Nearshore Research Group
University of Washington, Seattle, WA
(206) 543-7229
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r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html
Send "info&quo...
2005 Jun 07
1
Specifying medoids in PAM?
...w32
arch i386
os mingw32
system i386, mingw32
status
major 2
minor 0.1
year 2004
month 11
day 15
language R
Any guidance would be appreciated.
David
--
David Finlayson
Marine Geology & Geophysics
School of Oceanography
Box 357940
University of Washington
Seattle, WA 98195-7940
USA
Office: Marine Sciences Building, Room 112
Phone: (206) 616-9407
Web: http://students.washington.edu/dfinlays
2005 May 04
1
Calculate median from counts and values
...):
c( rep(16, 0), rep(8, 0), rep(4, 0), rep(2, 1), rep(1, 1), rep(0.5, 3),
rep(0.25, 16), rep(0.125, 55), rep(0.0625, 24))
I am sure that this is a one-liner for an R-master, but I can't figure it
out without a set of nested for loops iterating over each row in counts.
David
--
David Finlayson
Marine Geology & Geophysics
School of Oceanography
Box 357940
University of Washington
Seattle, WA 98195-7940
USA
Office: Marine Sciences Building, Room 112
Phone: (206) 616-9407
Web: http://students.washington.edu/dfinlays/
2000 Sep 19
1
on-line help failed for ultra 10 (solaris 8)
...rent working directory
/opt/sfw/R/R-1.1.1/src/library
*** Error code 1
make: Fatal error: Command failed for target `help'
Current working directory
/opt/sfw/R/R-1.1.1/src/library
*** Error code 1 (ignored)
Any help/advise for kick starting the help system
would be appreciated.
Thanks,
David Finlayson
=====
--
David Finlayson
david_finlayson at yahoo.com
University of Washington
Box 351310
Seattle, WA 98195 - 1310
__________________________________________________
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2005 Aug 26
1
Re: Speex-dev Digest, Vol 15, Issue 15
Seems to me that they are using speex already.
Ethereal shows that the voice RTP stream's payload type is 103. According to
this page [
http://www.openh323.org/pipermail/openh323/2004-June/068705.html], 103 is
SpeexNarrow-8k, although this email from Craig might not be correct.
- Cheng
> Message: 2
> Date: Thu, 25 Aug 2005 04:15:44 -0500 (CDT)
> From: Ashhar Farhan
2003 Sep 11
3
SIP to SIP monitor and record?
Hello All,
Is it possible to monitor and record a SIP to SIP call? If so, how?
I gathered from some previous posts this would not be possible.
--
Thanks,
Tim
2004 Aug 06
1
Speex SIP support in the "Asterisk" PBX, FYI
...Ogg or at least encode the size of each packet so the
>decoder knows how long it needs to read.
I'll probably ending up adding an option to output the data as an 'ogg'
format file (just as there's currently an option to output the data as a
'.mov' file).
Ross Finlayson
LIVE.COM
<http://www.live.com/>
<p><p><p>> Jean-Marc
>
>--
>Jean-Marc Valin, M.Sc.A.
>LABORIUS (http://www.gel.usherb.ca/laborius)
>Université de Sherbrooke, Québec, Canada
>
--- >8 ----
List archives: http://www.xiph.org/...
2002 May 16
3
Wind rose plots?
Just out of curiosity, has anyone created wind roses using R?
Thanks.
Tosh
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Send "info", "help", or "[un]subscribe"
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2003 Nov 14
2
Streaming channels from Asterisk to the Internet
Hi folks,
I'm wondering if it is currently possible to configure Asterisk to
forward the conversations from 2 channels into a streaming daemon,
such as Icecast, so that other people connected to the Internet can
listen.
The concept is similar to a radio talk-show. The show host would
connect to Asterisk via an X100P or through VOIP. His or her voice
would then be sent to the streaming
2004 Aug 06
2
Speex SIP support in the "Asterisk" PBX, FYI
FYI, the Asterisk software PBX <http://www.asterisk.org/> has now
incorporated my recent patches to support dynamic RTP payload types. As a
consequence, its SIP implementation now supports Speex, so if you have a
Speex-compatible SIP client, you can use it to make calls using Asterisk.
Some caveats:
- Only narrowband (8 kHz) Speex is currently supported; not
wideband. (Unfortunately,
2003 Mar 18
0
Ogg Traffic for March 18, 2003
...e specification contains errors that need
to be corrected. Having correct decoder specifications is essential
for a codec, and thanks to John's efforts, the Vorbis specs are that
much closer to being 100% correct.
4. Recent Developments
4.1. Asterisk PBX supports Speex
Ross Finlayson informs us in [6]this message that the software PBX
[7]Asterisk has recently accepted his patches to support dynamic
payload types, and as a consequence, Asterisk's SIP implementation now
supports Speex!
4.2. Speex heading for the RFC Track
Greg Herlein, main author of th...
2003 Oct 10
2
Vorbis-RTP situation
Is anyone working on Vorbis over RTP?
It's been silent on this frontier for a while. I have been hearing from
representatives of the IETF that they were not impressed by the previous
draft, and that they were not coherent with RTP thinking and other RTP
transport specs for various content.
There exist a textbook on RTP which was released recently, if someone
wants to learn it from the ground
2000 Nov 20
4
Vorbis over RTP
Hi.
I got bored of writing papers last week so I decided to write an app for
sending and receiving multicast vorbis streams over RTP. A first version
is available here:
http://www.cdt.luth.se/~rolle/vorbis/
(I use the JOrbis decoder, so you need Java 2 with Javasound, e.g.
JDK1.3. Since the decoding is done in Java, you probably need at least
300-350 MHz. Works with the IBM JDK1.3 on Linux on a
2000 Nov 20
4
Vorbis over RTP
Hi.
I got bored of writing papers last week so I decided to write an app for
sending and receiving multicast vorbis streams over RTP. A first version
is available here:
http://www.cdt.luth.se/~rolle/vorbis/
(I use the JOrbis decoder, so you need Java 2 with Javasound, e.g.
JDK1.3. Since the decoding is done in Java, you probably need at least
300-350 MHz. Works with the IBM JDK1.3 on Linux on a
2004 Aug 06
6
*Real* real time streaming (no delay/latency)?
Hello
Does anyone have experience with _real_ real time streaming - i.e. with
very little delay/latency?
I need to stream from point A to point B in near-CD quality via a 100
Mbit network. That is easily done using icecast. But here is the tricky bit:
I want as little delay in the signal as possibble - preferrably below 50ms!
I have made a test setup encoding on and serving from an 800 MHz
2004 Aug 06
3
Updated Speex RTP Internet Draft
...lso like to thank Brian
C. Wiles <brian@streamcomm.com> of StreamComm for his assistance in
developing the proposed standard for Speex use in H.323
applications.
The authors would also like to thank the following members of the
Speex and AVT communities for their input: Ross Finlayson,
Federico Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.
<p>10. Author's Address
Greg Herlein <gherlein@herlein.com>
2034 Filbert Street
San Francisco, CA
United States 94123
Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>
Departmen...
2007 Jun 07
1
draft-ietf-avt-rtp-speex-01.txt
...ian C. Wiles
> <brian@streamcomm.com> of StreamComm for his assistance in developing
> the proposed standard for Speex use in H.323 applications.
>
> The authors would also like to thank the following members of the
> Speex and AVT communities for their input: Ross Finlayson, Federico
> Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.
>
> Thanks to former authors of this document; Simon Morlat, Roger
> Hardiman, Phil Kerr.
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&g...
2007 May 29
0
draft-ietf-avt-rtp-speex-01.txt
...uld also like to thank Brian C. Wiles
<brian@streamcomm.com> of StreamComm for his assistance in developing
the proposed standard for Speex use in H.323 applications.
The authors would also like to thank the following members of the
Speex and AVT communities for their input: Ross Finlayson, Federico
Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.
Thanks to former authors of this document; Simon Morlat, Roger
Hardiman, Phil Kerr.
Herlein, et al. Expires November 30, 2007 [Page 17]
Internet-Draft...
2007 May 30
5
draft-ietf-avt-rtp-speex-01.txt
Do not forget to add the "Copying conditions" to the RFC.
Check http://wiki.debian.org/NonFreeIETFDocuments
That page contains a section titled "Template for RFC authors to
release additional rights". To follow that guideline a
section like the following should be added:
x. Copying conditions
The author(s) agree to grant third parties the irrevocable
right to
2004 Aug 06
0
draft-herlein-speex-rtp-profile-01
...lso like to thank Brian
C. Wiles <brian@streamcomm.com> of StreamComm for his assistance in
developing the proposed standard for Speex use in H.323
applications.
The authors would also like to thank the following members of the
Speex and AVT communities for their input: Ross Finlayson,
Federico Montesino Pouzols, Henning Schulzrinne, Magnus Westerlund.
<p>10. Author's Address
Greg Herlein <gherlein@herlein.com>
2034 Filbert Street
San Francisco, CA
United States 94123
Jean-Marc Valin <jean-marc.valin@hermes.usherb.ca>
Departmen...