similar to: codec for really low bitrates

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

Hi, i have a problem with Guild wars, it runs only in the backround. I ran it through terminal and got this: Code: fixme:win:EnumDisplayDevicesW ((null),0,0x32eb54,0x00000000), stub! fixme:win:EnumDisplayDevicesW ((null),0,0x32e6c0,0x00000000), stub! fixme:devenum:DEVENUM_ICreateDevEnum_CreateClassEnumerator Category {cc7bfb41-f175-11d1-a392-00e0291f3959} not found

You might wish to read this thread as well, for some backround on LLVM IR. http://lists.cs.uiuc.edu/pipermail/llvmdev/2011-October/043719.html Summary: LLVM IR is target specific, not portable between different targets. LLVM IR is actually a Compiler IR and not a virtual machine language.

Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an

hello, i just got introduced to R - WOW its beautiful.. I am presently a SAS user and wanted to configure R to work in a multi-user enteprise environment. Client - Server. Where we have a strong LINUX server supporting about 10 statisticians with R. Anyone have any backround or information they can share to help me get jump-started on setting up R in this environment? Does each user have

Hello, I've installed R on my RHEL3 cluster and I am trying to get Rmpi to work properly. R is installed using the following ./configure --prefix=/home/apps/R-2.2.0 I installed snow using R CMD INSTALL /home/apps/snow And finaly Rmpi R CMD INSTALL /home/apps/Rmpi --configure-args=--with-mpi=/path/to/lam There were no errors or warnings upon installation. However when i perform the test

Hi, I wonder if it's possible to use vorbis for streaming audio (live). How is the speed in comparison to lame. How is the quality with low bitrates (16k-32k). If quality and speed is better than lame it should be possible to use it with a icecast server for live streaming (with some modifications). Sound quality is very bad with low bitrate mp3, realaudio is much better, but I don't like

Hi all, ome people asked if it was possible to make an even higher quality mode by incorporating some of the changes from the first tuned version into the 350kbps mode. I did so and made a new version with this new mode. It gives bitrates from roughly 300-350kbps. That's a lot, but it also gets very hard to find something it artifacts on :) Should be sufficient for archival quality. In

I'm seeing some rather odd behavior with respect to the match percent that is returned when performing a simple query. It appears that the addition of a filter to a simple OR query will not affect the weight of the match, but will increase the percent of the resulting match. According to the documentation for MSetIterator::get_percent (

Hello all, I'm glad to announce to you that Vorbis now has full ReplayGain support. If you're not familiar with ReplayGain, take a look at www.replaygain.org. The main features are: a) all songs play back with equal loudness b) removes the need for normalization c) allows for clipping prevention Using it is very simple. Get a compatible decoder (ogg123, XMMS and WinAmp all support it

Hello, HawkVoice doesn't have a 6.3kbps codec for CELP, it has a 4.5kbps CELP codec and I do not believe it is being used by TeamSpeak. The 6.4kbps CELP being used in TeamSpeak, to which you are referring I believe comes from Lernout & Hauspie's LHACM.ACM file which it appears you are redistributing (I assume TeamSpeak has a license and permission to do this). The only people I

hello celt developers. im just wondering if its possible to compile celt codec for windows? if so, could someone point me in the right direction? thanks.

In data OME in MASS I would like to extract the first 5 observations per subject (=ID). So I do library(MASS) OMEsub <- split(OME, OME$ID) OMEsub <- lapply(OMEsub,function(x)x[1:5,]) unsplit(OMEsub, OME$ID) - which results in [[1]] [1] 1 1 1 1 1 [[2]] [1] 30 30 30 30 30 [[3]] [1] low low low low low Levels: N/A high low [[4]] [1] 35 35 40 40 45 [[5]] [1] coherent incoherent coherent

Hi, I want to encode audio at very low bitrates. however, vorbis_encode_init() does not allow me to set bitrates below 32kbps. How can i set 24kbps? Thanks, Flo --- >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 'vorbis-dev-request@xiph.org' containing only the word

Hi, I'm really happy Ogg Vorbis 1.0 is out. I have some questions though: - what are the valid bitrates / channel / sample rate combinations? experimenting I find that at 22050 Hz I can not use 24 kb/s stereo, and at 16000 Hz I cannot use 18 kb/s stereo. Is there a table of valid bitrate / sample rate combinations? <p>- is there documentation for the encoding API? looking at the

On Aug 09 11:58:33, u.windl at ukr.de wrote: > > -----Original Message----- > > From: opus <opus-bounces at xiph.org> On Behalf Of Jan Stary > > Sent: Friday, August 9, 2024 12:00 PM > > To: Petr Pa??zek <petrparizek2000 at yahoo.com> > > Cc: opus at xiph.org > > Subject: [EXT] Re: [opus] Re: Opus Tools -- low bitrates, new features in 1.5, > >

On Fri, Oct 10, 2014 at 07:32:06PM -0600, Eric Blake wrote: > On 10/10/2014 11:37 AM, Jd wrote: > > Hi > > Looking in to implementing (CBT like) delta backup for KVM. > > Not quite sure what you mean by CBT. > > > > > The following looks promising..(last paragraph) > > http://wiki.qemu.org/Features/Snapshots2 > > > > Libvirt

Dear Vorbis Devteam, My name is Michael Feilen and I've been working on Digital Radio Mondiale (DRM) transmitters and receivers for quite a while now. DRM uses HE-AACv2 by Dolby to encode the audio content (see http://www.drm.org/uploads/media/es_201980v030101p.pdf - pages 23 ff). As I think Vorbis is an excellent alternative, I'd like to implement and define an interface for Vorbis

> What sine sweep exactly? An exponential sweep. It started slightly below 24 Hz and ended almost at 24 kHz. And it was 50 seconds long. > How did you obtain it, I used Angelo Farina's "Aurora" modules. One of them is called "Generate sine sweep". > and how exactly did you encode and decode it? 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav

> To be clear: did you mean the opus output of opusenc > or the wav output of opusdec? I meant during the decoding. There's one significant difference between how Opusdec deals with resampling and how, let's say, MP3 decoders usually deal with resampling. If I make an MP3 at a very low bitrate and if the encoder decides (because it's too low) to internally resample my audio