similar to: Help with bad audio using MPC..

flight 8091 xen-unstable real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/8091/ Regressions :-( Tests which did not succeed and are blocking: test-amd64-i386-rhel6hvm-intel 12 guest-localmigrate/x10 fail REGR. vs. 8071 Tests which did not succeed, but are not blocking, including regressions (tests previously passed) regarded as allowable: test-amd64-amd64-xl-pcipt-intel 9

Signed-off-by: Ilia Mirkin <imirkin at alum.mit.edu> --- drivers/gpu/drm/nouveau/core/engine/graph/nv50.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/drivers/gpu/drm/nouveau/core/engine/graph/nv50.c b/drivers/gpu/drm/nouveau/core/engine/graph/nv50.c index 0f8d18a..30ed19c 100644 --- a/drivers/gpu/drm/nouveau/core/engine/graph/nv50.c +++

Hi guys. First off, keep up the good work on FLAC: great now, and looks promising in the future :) I wonder if anyone's aware of the issues with FLAC->mpc transcoding: it works under linux but not windows. Sorry if this has been bought up loads before - i couldn't see it here. Here's the message mppenc 1.02 comes up with on trying to encode a .flac : "MPC Encoder 1.02

I am very happy to announce that the aoTuV tuning of Ogg Vorbis has tied with Musepack at first place in the 128 kbps listening test. It has beat iTunes AAC, Lame MP3, ATRAC3, and WMA standard. :) http://www.rjamorim.com/test/multiformat128/results.html -- ------------------------------------------------- Stephen So PhD Student Signal Processing Laboratory School of Microelectronic

128 kbps! In-Reply-To: <E77B14D5-B72A-11D8-91A8-000A95A4DC02@kernel.crashing.org> Message-ID: <Pine.LNX.4.44.0406081019270.6447-100000@gorlois.cs.upb.de> On Sat, 5 Jun 2004, Segher Boessenkool wrote: > There are quite a few models of how the ear works. All the good > ones are computationally expensive, and not usable at all > mathematically. Your paper uses just one simple

In researching Ogg Vorbis, I have started to become aware of many other audio formats I never knew about, such as FLAK. Well, today I have become aware of two other formats I hadn't heard of before: MPC and AAC. Can anyone tell me what they think of these formats in comparison to Ogg Vorbis? If anyone considers this Off-Topic, please feel free to email me personally. I'm not trying to

Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

I hacked around a bit with the simple case of tracing just classes and defs (no multiclasses or defms). Below you will see my test file and then the output produced. Note that the regular output from the PrintRecords backend follows the trace, so you can see the final classes and records there. Once the trace can be selective, it makes sense to add another option for PrintRecords that restricts

I found that machines running CentOS 6.6, kernel 2.6.32 and Samba 4.4 maps UID to values under 16 bits for instance: uid=12112(john) gid=100(users) groups=10102(DomainUsers) however other systems parts of the same AD but running CentOS 7 (kernel 3.10 and Samba 4.7) use different, much larger IDs, for instance: uid=10499212112(john) gid=100(users) groups=10102(DomainUsers) But with a similar

I'm interested in computing densities of a number of small ego-centred or local networks. I've installed both the R program and the sna package. My next task is importing the data into R. Our data was entered into Spss in an edgelist format in which each case specifies the link between an alter and an ego. For example: 10101 10102 1 10101 10103 3 10102 10104 2 10201 10203

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

Hello, All. I am currently running a test configuration for the telephony engineer here. I have two Dell PE servers, each with 2 A108 Sangoma T1/E1 cards. Here is a rough drawing: Ameritec Call Generator 8 T1 ISDN/PRI lines ---> Asterisk ---> Asterisk ---> Ameritec Call Generator Sangoma A108 Sangoma A108 Terminating 8 ISDN

Hi All, I have this codec problem, I use "gsm" in my iax.conf file and in teliax settings also, but the error is still appearing as below. please help me. Kumara Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by

Hello, After commit 93af05e03e05d2f85b5a7e20ec3a3a543584d84f we have new warning but only if compiled with GCC: In file included from /path/to/llvm/include/llvm/ExecutionEngine/Orc/ExecutionUtils.h:19:0, from /path/to/llvm/lib/ExecutionEngine/Orc/OrcMCJITReplacement.h:23, from /path/to/llvm/lib/ExecutionEngine/Orc/OrcMCJITReplacement.cpp:9:

While ripping one of my cds on my laptop this happened: Message from syslogd@theirongiant at Thu Jul 5 09:52:16 2001 ... theirongiant kernel: Assertion failure in do_get_write_access() at transaction.c line 551: "handle->h_buffer_credits > 0" from the kern.log: Assertion failure in do_get_write_access() at transaction.c line 551: "handle->h_buffer_credits > 0"

How configure openvpn tunnel in shorewall with differents src and dst ports ? openvpn.conf ... lport 10101 rport 10102 ... Leandro.

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

Hello I'm trying to install Dahdi through source code on a Fedora 13 host to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv isn't happy. 1. After successfully running "make all; make install; make config", I edited /etc/dahdi/system.conf thusly: loadzone=fr defaultzone=fr fxsks=1 2. Then ran "dahdi_cfg -vv" which says: ------------- DAHDI Tools

Hi all, yesterday afternoon, I called through my provider (teliax). but from the evening, I get this error. (below). then I checked in My Account page ans support page in teliax. and I saw that they have given new setting (to another proxy sever). I followed new settings. my Asterisk server is connecting to the teliax. but still I con not make called. it shows this error. If somebody had this