similar to: YA-2496

Hi people, looking around for a new audiocard, my eye fell on the M-audio audiophile 2496. It has 4 digital in/out and is 24bit, 96kHz. The sound quality is very good, if I can believe the reviews. <p>My question is: can vorbis do 24bit, 96kHz ? -- --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list,

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for

I recently bought a DVD Audio disc and I searched the web and wasn't really able to find much about DVD Audio and Linux. I have a few questions that I was hoping some people on this list might be able to answer. How can I extract the uncompressed DVD Audio (not the DTS Audio portion of the disc, but the 96khz/24bit/5.1 pcm part)?

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and

Hi I get the following warings on my ubuntu (gutsy) machine > library(matrix) > data.frame(x=1:5, z=5)[1:5, ] x z 1 1 5 2 2 5 3 3 5 4 4 5 5 5 5 Warning message: ':::' is deprecated. 'methods' is deprecated. 'seemsS4Object' is deprecated. Use 'isS4' instead. See help("Deprecated") It is quite annoying since I often subset the data.frames. Will

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

Hi all, i'd like to ask what the best options are for safe 24bit 96kHz archiving. Currently i'm only using -8 but there are also some other options like block size etc. Can anyone suggest? Also i'd like to ask whether a no-lax 192kHz mode is planned in flac, seems like 192kHz isn't directly supported although flac can compress such rate in lax mode. My last question - is there

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

Begin forwarded message: Date: Sat, 26 Sep 2015 13:46:13 -0700 From: John La Grou <jl at jps.net> To: Erik de Castro Lopo <erikd at mega-nerd.com> Subject: 32 bit data Erik, could you please post this to the FLAC mailing list. I've tried a couple times, but nothing happens. Tks. ......................... There are reasons to support 32-bit data. The emerging network standard

We have received an email in Ubuntu alerting us that something strange was going on during the r-base 2.7.1 rc build (see [1]). It seems that there is an error during the convertion of R-FAQ.texi to pdf. The following excerpt, taken from [2], shows the problem: /bin/sh: line 0: test: too many arguments TEXINPUTS=".:$TEXINPUTS" /usr/bin/texi2dvi --texinfo="@set UseExternalXrefs

I've just installed RC2 and I'm very excited about the quality. It's so much better than MP3. This is the first version I've used since I just found out about Ogg Vorbis. I did notice that very high frequencies seem to be missing but since not many people can hear much above 18 KHz it's not much of an issue. I suppose this resolves the hiss problem so prevalent in MP3.

Seems time is going to be a constraint for me in the coming week, but I'll try and get more of these out to you as fast as I can get them written and tested. Keeping the patches in a separate branch for now seems like a good idea to me. Attached is a patch for 'test_streams.sh' On Tue, Mar 12, 2013 at 1:44 AM, Erik de Castro Lopo <mle+la at mega-nerd.com>wrote: > Jaren

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

Hi everyone, I am looking to do some manual maximum likelihood estimation in R. I have done a lot of work in Stata and so I have been using output comparisons to get a handle on what is happening. I estimated a simple linear model in R with lm() and also my own maximum likelihood program. I then compared the output with Stata. Two things jumped out at me. Firstly, in Stata my coefficient

The results are at www.rational.co.za/speex.csv Each of the 11 quality settings is tested 3 times (narrow, wide and ultra wide band). Strangely narrow band quality 11 outperforms all wide band tests, but it can be due to my subsampling or some other inaudible effect like delaying. You can import it into Excel and sort it by SNR or other value. Divide the bits by 24 to get the bps. The

Author: motu Repository: /hg/zfs-crypto/gate Revision: cf616258f6c9e546db6d1ed74f00e434f369e5e7 Log message: 6222529 obsolete serd properties are not reported as illegal Files: update: usr/src/cmd/fm/eversholt/common/check.c

Could you use something like a MOTU firewire box to input all your streams into one computer? They have lots of audio inputs. I don't know anything about the practicalities I'm afraid, so no idea how to turn those inputs into separate streams. But I'd much rather do that than run 8 separate computers. Dan On 19/05/05, Seth McTigh <kleptein@hotmail.com> wrote: > I have a

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low