similar to: raw_bits branch

From: Thorvald Natvig <slicer at users.sourceforge.net> --- celt.pc.in | 2 +- configure.ac | 2 ++ libcelt/Makefile.am | 12 ++++++------ tests/Makefile.am | 2 +- tools/Makefile.am | 4 ++-- 5 files changed, 12 insertions(+), 10 deletions(-) diff --git a/celt.pc.in b/celt.pc.in index 98cc008..67a830b 100644 --- a/celt.pc.in +++ b/celt.pc.in @@ -10,5

--- libcelt/Makefile.am | 6 +- libcelt/bands.c | 26 +++++++++- libcelt/bands.h | 2 +- libcelt/celt.c | 23 +++----- libcelt/pgain_table.h | 133 ------------------------------------------------- libcelt/quant_pitch.c | 117 ------------------------------------------- libcelt/quant_pitch.h | 44 ---------------- 7 files changed, 37 insertions(+), 314

--- configure.ac | 7 +++++ src/libFLAC/bitreader.c | 12 ++------- src/libFLAC/bitwriter.c | 8 ++---- src/libFLAC/fixed.c | 18 +++++-------- src/libFLAC/format.c | 8 ++---- src/libFLAC/include/private/macros.h | 29 ++++++++++++++++++++ src/libFLAC/metadata_iterators.c | 17 +++---------

I changed just the FLAC__EntropyCodingMethod_PartitionedRice struct, but it hasn't changed a thing. I'm still getting the same behaviour. typedef struct { unsigned order; unsigned parameters[1 << 6]; unsigned raw_bits[1 << 6]; } FLAC__EntropyCodingMethod_PartitionedRice; Any other thoughts? Thanks, Reza Josh Coalson wrote: > --- Reza Naima

On Wed, 4 May 2016, Nux! wrote: > Direct links > > https://www.imagemagick.org/discourse-server/viewtopic.php?f=4&t=29588#p132726 > https://bugzilla.redhat.com/show_bug.cgi?id=CVE-2016-3714 > > Mitigation: > > As a workaround the /etc/ImageMagick/policy.xml file can be edited to disable > processing of MVG, HTTPS, EPHEMERAL and MSL commands within image files,

hi everybody, we have problems using wincvs with a cvsroot on a samba-server. we think that the problem is the renaming of files where the cvs user does not have write-permisson on: this is our configuration: fileserver on a irix6.5 running samba 1.9.18p8 having a share on / named 'coder'. win95 box, running wincvs, setting cvsroot something like

On 05/04/2016 08:15 AM, John Hodrien wrote: > On Wed, 4 May 2016, Nux! wrote: > >> Direct links >> >> https://www.imagemagick.org/discourse-server/viewtopic.php?f=4&t=29588#p132726 >> >> https://bugzilla.redhat.com/show_bug.cgi?id=CVE-2016-3714 >> >> Mitigation: >> >> As a workaround the /etc/ImageMagick/policy.xml file can be edited

On Sun, Jun 24, 2001 at 02:30:56PM -0700, Josh Coalson wrote: > There have been reports of -9 using huge amounts of memory. -9 is really > theoretical, but people always seem to want to try the max setting. Anyway, > that's not an excuse but figuring out why -9 is using so much memory is lower > on my list than other stuff. -8 should get within 0.01% of -9 and is pretty >

Hi, ? Is there a fix for this issue??? ---> http://permalink.gmane.org/gmane.comp.audio.compression.speex.devel/2959 ? I am seeing the same thing when I compile speex in visualdsp ? These are the errors I get from using the assembly version of vq_nbest: ? ..\..\..\..\algorithms\voice\speex\src\vq.c [Error ea5004] "C:\Users\coder\AppData\Local\Temp\acc22e8547f000\acc22e8547f001.s":482

I know it's conventional to report bugs to the maintainer, but I'm not sure which package actually contains this bug(s), so I apologize for sending this to the list at large. I see the bug under both R 2.1.1, and R 2.2.1. (I sent a related message a while ago, but this one has more detail.) library(MASS) library(nlme) fit.model <- function(il, model.family) { cs <-

I am trying to reshape a data set. Could someone please help me with the reshape, cast, and melt functions? I am new to R and I have tried reading up on how to use the reshape package, but I am very confused. Here is an example of what I am trying to do: subject coder score time [1,] 1 1 20 5 [2,] 1 2 30 4 [3,] 2 3 10 10 [4,] 2 2

Hi, I've trying to decode a FLAC audio stream. I have a reader which sends raw byte data to my FLAC wrapper class. Only once the decode function below returns, the reader will send new data. Hence I want to decode until the stream is empty, but I will add new data to the stream once it is empty. *void **MyFlacCoder::**decode(char *data, int bytes) { mBuffer = input;

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Hello, I have a rather annoying issue on going with one of my centos virtual servers. the server hosts a website using apache and mysql ,there are three persons involved with keeping the site up and running. and i am his root due to the fact he does not know anything with about Linux. there is an php/sql coder , and the site owner which only knows to use the CMS and upload new articles to the

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Steve, My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about my findings here, This behavior can be reproduced. But '*' do not seem to do the negotiation correctly. http://lists.digium.com/pipermail/asterisk-users/2004-January/032197.html

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

I've come into some code that produces different results under R 2.1.1 and R 2.2.1. I'm really unfamiliar with the libraries in question (MASS and nlme), so I don't know if this is a bug in my code, or a regression in R. If it's a bug on my end, I'd appreciate any advice on potential causes and relevant documentation. The code:

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

Hello, I am a computer engineering student and compression hobbyst and have recently developed a new entropy coding algorithm. It can be used to achieve compression proper of arithmetic coders at very high speeds -almost like Huffman codecs-. Since it could be of interest to you, I send it as an attachment -code and technical report-. Please, drop me a line in case you have any doubt or