similar to: DVD Audio & Vorbis

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,

I've been reading through a few forums about ripping DVD audio / Sony SACD's (I'm not sure if these are the same or if they are different formats). Anyway, being that I haven't found a way to rip this stuff yet as there may not be any consumer product out yet that enables the feat, I have been curous about FLAC's ability to encode in a 6 channel format at the high bitrate

On Sun, 23 May 2004, Conrad Parker wrote: > On Sun, May 23, 2004 at 07:04:28AM +1000, Kenji Chan wrote: > > Is that ok for vorbis to encode 24bits/96KHz audio? > yes, oggenc should be able to handle it. Not only does it handle it fine: but most players will downsample the 96k audio for playback on non 96k able hardware. (24bittage isn't an issue as Vorbis is 24bit internally,

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

On Fri, Jun 20, 2014 at 01:21:03PM +0400, lvqcl wrote: > Miroslav Lichvar ?????: > > > +/* > > + * This is used to avoid overflow with unusual signals in 32-bit > > + * accumulator in the *precompute_partition_info_sums_* functions. > > + */ > > +#define FLAC__MAX_EXTRA_RESIDUAL_BPS 4 > > > + /* WATCHOUT: "+ bps +

I have a Debian Sarge box that I am trying to get icecast2 and ices2 up and running on. I'm having problems, but do not know where to look to identify the problem. I have my audio working with ALSA - microphone in plays through my audio out. I'm running a basic configuration using the defaults provided, and the ALSA example config file for ices. When I try to connect to the server using

Matt Boersma wrote: > This is slightly off-topic, but since we're talking about sample rates > perhaps one of you can help. > > I have a SoundBlaster Live! 5.1 card installed in a nice server > running Debian Woody (stable). I have ices and icecast2 working > well from recent CVS sources. Linux 2.4.18 kernel using the included > OSS/Free emu10k1 driver. > >

ARGH! I am at a complete loss as to which OGG quality settings to use: 8? 10? 3? I'd like to be able to listen to my primarily Rock oriented music on a high-end system (though I don't own one - yet) without any noticeable sound degradation, but I don't want to go total overkill with -q 10. With LAME, I at least used to know 192 kbps with -q0 was a perfect size/quality proportion. I

wow! i must say vorbis deprecates mp3! lol. although, i had to use the "OSS/3d for XMMS" plugin on my sblive value w/ 4 point sound to get a proper base. On my other box with a diff card, bass is just fine. however, are there any h/w players for vorbis files? the faq only states one manufacturer called iObjects but i see no portable players! Basically, i want to player vorbis files

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

Hi. I've been browsing the archive on this topic and only found a few notes, all dating from a year ago (almost too precisely :) ) -- hope I haven't skipped the mails on that matter, sorry if I did. <p>Basically, I will get in the next few months a MOTU 896, that have 8 i/os in 24/96. I do pro sound recording, so it's more or less my business to have such a piece. Of

Hi Guys, I've just faced a wrong warning trying to encode a 24 bits WAV file if(wFormatTag == 1) { if(bps != 8 && bps != 16) { if(bps == 24 || bps == 32) { /* let these slide with a warning since they're unambiguous */ flac__utils_printf(stderr, 1, "%s: WARNING: legacy WAVE file has

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hello to all. I have a VERY important correction to this article. To compile wine on Debian Etch amd64 (kernel 2.6.17) you'll need some changes: 1. You don't need to make the X11R6 symlinks 2. The package "ia32-libs-dev" does not longer exist, you need libc6-dev-i386 3. make and make depend have to be run with LD_LIBRARY_PATH="/lib32" I hope now it is all regards,

Hi everyone, Over the last two weeks or so, I've been thinking about how to add surround sound to Ogg -- and more than that, to do it in the best way possible. With this in mind, I started considering using Ambisonic surround sound. The advantages of this format are considerable: a) It was developed in the early to mid '70s, so the patents should be expired by now.

Hi, I am a developer from Oracle. I wanted to create a png file out of png device with a bit depth of 8. I am using R on linux. But I noticed that R automatically switches between 8 and 24 bit depth. i tried a lot of things from your grDevices package but unable to control the bitdepth. I would be really grateful to you guys if you can suggest me some way to solve this issue. -/Ag Anshul Gupta

Hi all, I have the Icecast2 server running perfectly and the iceS streamer feeding it either /dev/dsp or a playlist of ogg files, depending on which config I run iceS with (I have three config files: one for live, one for music, and one default (current)). I'm using Slackware w/the 2.4.22 kernel. The problem I'm having is perhaps theoretical: how can I talk a little, play some

I've been playing with speex for use in a VoIP application between PC's. One thing I've found (correlating to the documentation) it that speex runs much faster and produced much better output when it's fed a 32kHz signal instead of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the

Hi list! I am currently investigating the possibility of allowing musicians to collaborate over the internet -- Musician A -------> internet --------> Musician B This diagram obscures quite a lot of detail. In reality, the datapath would look more like this: Musician A ADC Compression algorithm UTP (or perhaps RTP?) IP ADSL the Internet ....<snip>..... Clearly, this link would

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re: