similar to: Why Point-Stereo at 160 kbps ?

I have some problem whit Kerberos. OS: FreeBSD 5.3 Domain: W2k3 native mode. 1)I am Installing Heimdal 0.6.1 over port. Config /etc/krb5.conf %/usr/local/bin/kinit ivan ivan@NKMK.RU's Password: kinit: krb5_get_init_creds: Response too big for UDP, retry with TCP 2)Compile and install Heimdal 0.6.4 over source %/usr/local/bin/kinit ivan ivan@NKMK.RU's Password: kinit:

I noticed that my previous message is not very complete so I send here an "enhanced version". Please disregard the old one an reply to this one only. ( you can delete the ATTENTION word from subject ) Wilson (defiler@null.net) wrote : > There are two ways to decode multi-channel audio. In hardware, or in > software. > Hardware: A receiver or processor takes a Dolby Digital

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

I halved the volume of the sample before encoding with `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. So the artifact I heard was just clipping when encoding or decoding. Opus remains unbeaten for me at 133 kbps. That's totally awesome. Cheers On 10 April 2017 at 17:14, Agustín Dall'Alba <agustin at dallalba.com.ar> wrote: > Hello! I found a

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > I halved the volume of the sample before encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding

Hello! I found a sample I can ABX successfully when encoded at 133.333 kbps. I was targetting 1 MB/min. https://drive.google.com/drive/folders/0B8KWShoIrA1kQzR1Z0FFRUlfcEU floex.wav is 4:54–5:04 of a lossless copy of 'Forget-me-not' by Floex, downloaded from http://store.floex.cz/album/zorya floex-133.opus was created with `opusenc --bitrate 133.333333 floex.wav floex-133.opus`,

> > So > > it's closer to Dolby AC-X than to VQ. Unfortunately, I'm not sure that it's > > not patended by Dolby. > > Could you be more specific? More than anything else, we need to know more > about the patent waters we're navigating. > > Monty As Dolby AC-X is using enveloppe of the sound since several years, and as it seems that Dolby was

----- Forwarded message from Mark Lewis <mlewis@webnoize.com> ----- Delivery-Date: Tue Nov 21 10:15:55 2000 Date: Tue, 21 Nov 2000 10:07:15 -0800 From: Mark Lewis <mlewis@webnoize.com> Reply-To: mlewis@webnoize.com Organization: Webnoize News X-Mailer: Mozilla 4.75 [en] (Win98; U) To: monty@xiph.org Subject: liked your article at http://xiph.org/about.html here's an article about

Well, in the world of audio compression it's never boring... :-) Full press release from Dolby: http://www.aac-audio.com/press/aac.pr.0104.AACencoder.html My comments: Looks like Dolby will start licensing AAC encoder that targets consumers and the products they use like CD rippers, jukebox players, etc. Also, it looks like that this encoder will be Low Complexity AAC ("The AAC Consumer

On 20030610: Gregory Maxwell wrote: (in reply to Ralph Giles) >> I assume you're aware of the technical documentation on dolby's site? >> (http://www.dolby.com/pro/) In particular the surround mixing guide has >> a lot of detailed guidelines. I don't have any practical experience >> with it though, so I can't vouch for it. >Yes I am. I've done a

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

On Fri, Nov 11, 2005 at 09:13:01AM +1100, Erik de Castro Lopo wrote: > > However, support for (ie) 48-bit-float should not have to be created, > > Where are you going to find a 48 bit float? Is there an IEEE > standard for that? It's not about what's now, it's about what could be, and nobody has been able to make good predictions, except your logic here: > If

Marvin, fair enough. Via's licensing terms appear to apply to encoders and decoders only, and not to transports, such as Icecast ("An AAC patent license is needed by manufacturers or developers of end-user encoder and/or decoder products"), so using Icecast to transport an AAC-encoded stream is not a violation of the AAC is not mentioned in the Icecast documentation (that I can

Arc wrote: > Ok so we cap it to 64bit, since much more than that doesn't make sense (96bit > would be a "long double" C type) On x86 CPUs, "long double" is 80 bits. > I really don't like this idea, but I will entertain, formatting it as follows: > > ID Type Bits > 0 Int 8 > 1 uInt 8 > 2 Int 16 > 3 Int 24 > 4 Int

On Fri, Nov 11, 2005 at 07:17:53PM +1100, Erik de Castro Lopo wrote: > We're talking about a file header here. Even if the header is a kilobyte in > size, it will be completely **dwarfed** by the audio data following. So why > are you counting single bits like this? Why waste? You only have to read the header once for a stream, and libogg2 provides a convient bitpacker which can

I must say this is a very confusing issue. I generate AAC+ on an open source system which downloaded some code during installation from the 3GPP source. It appears the encoding software is being freely distributed, and hardware which decodes the stream pays a licence fee at point of manufacture. We have such hardware. In a number of locations. One brand has AAC+ only, not AAC. I note VLC

As far as I know, LAME is able to mantain the informations stored in some audio files in the lower frequencies, useful for a Dolby pro logic decoder to reproduce a surround sound. Is ogg able to do it? Olaf <olaf@ kjws.com> for every kind of mail, except spam! :-) --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

I am not a lawyer, information in this email is no legal advice! On 17 May 2017, at 2:09, Greg Ogonowski wrote: > It's really pretty simple. > You can download the code and build it all you want... ...for > yourself. > It cannot be distributed, sold, or used commercially in any way. > That's all. > /g. That only applies to the source code. You still need to license

First off, I'm not an expert in this field, but I thought I could probably answer a few of your questions. If anyone else wants to give a more complete explanation, please do. Brian Willoughby wrote: > Hi folks, > > I'm trying to get a handle on the many lossless audio formats from > the perspective of a software developer. I love the FLAC format > because it is open

ICecast i capable of sorts of streams, that is not your problem. Your stream source client (DSP) is the part who must be capable of streaming the format you want. For AAC use oddcast DSP www.oddsock.org capabale of AAC, LAME Mp3, and OGG (Free to use) very good at OGG specially with the vorbis 1.1 aoTuVb3 DLLs For AAC Plus (HE_AAC) use Orban opticodec for PC (witch is capable of every bitrate