similar to: Ogg streaming on low bandwidth

Hello, I'm seeking advice here as I'm intend in the not to distant future to start encoding my CD collection, and I'm not sure what to use. Do I use the still current v1.0, or am I better off using a newer build? Also, I've started to notice mention of versions 1.0.1 and 1.1 in these mailing lists. Are these worth holding out for? I'll probably encode at about Q7, and am using

Dropping the other topic as requested, I'd like to follow something up in a new thread. I want to throw a question or two out there and see what responses people give (if any ;-). Assuming that you think Vorbis could have developed faster than it has done in the past year, what in your opinion has been holding it back and what remedies do you suggest? Speaking for myself, I think that it

Hi, Has there been any progress on the ACM for FLAC that was being looked at by Steve Lhomme? Thanks, Owen.

Hello. First of all, thanx for that great codec. I would encode all my music with Ogg Vorbis in the future, if it hadn't been for some samples I encoded for test purposes: Everything was great as long a I used "natural" sounds/music, but certain pieces of synthesized music (specially techno) contained very audible artifacts even at high bitrates. So I just went on using mp3 for

Josh Coalson wrote: > --- engdev <engdev@ozemail.com.au> wrote: > >>Hi, >> >>Has there been any progress on the ACM for FLAC >>that was being looked at by Steve Lhomme? > > > Haven't heard anything here about it for a long time. Sorry, I sent my reply to engdev privately (because there's no good reply to). In short I haven't worked on

hello there, I just started using ogg files. And want to thank you all for this new format. I have written an addon for mirc (chatclient for irc chat networks) to play and exchange soundfiles and I am almost done with implementing the ogg vorbis format into this application. The mirc-scriptinglanguage provides me a command to read a specified number of bytes starting at a specified location of

I'm going to encode ~2000 CDs soon. All genres, but 90% of it has distorted guitars... Everything from punkrock to metal to industrial to goth to synthpop to classical to techno to whatever... I've heard that RC2 has some hearable artifacts, even in 192/256 kbps... There have been quite a few "bugreports" since RC2 with people sending samples that even I can differ from the

Hello: Does anyone know if Garf's tunings have been integrated into CVS yet? I am waiting for that before upgrading. Charles -- "Problem solving under linux has never been the circus that it is under AIX." (By Pete Ehlke in comp.unix.aix) -------------- next part -------------- A non-text attachment was scrubbed... Name: part Type: application/pgp-signature Size: 190 bytes

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

My icecast server (1.3.11) has been getting 404 errors for the past couple of days from the yp.shoutcast.com server. If I send the URL manually by telneting to the webserver at yp.shoutcast.com I get the following response: [chrisp@ayeka httpd]$ telnet yp.shoutcast.com 80 Trying 205.188.234.56... Connected to scastlb1.shoutcast.com. Escape character is '^]'. GET

Q: Is there any testing against a collection of known "hard-to-encode" clips before new releases? It would be an obvious thing, if you want to be serious about quality. I brought this up because I tried latest cvs version of oggenc on one of these standard clips I have. It's a 6 sec long clip of an applause. Heavy noise is easy to hear with qualities 0 to 5,99. (This corresponds

It's technically good at streaming for broadcasters because, if it encodes something simple, it drops the bitrate and therefore our costs. And it sounds great too, but you wanted technical reasons... ! By the way, Virgin Radio UK's broadband streams are now proper stereo, instead of the suspicious mono-sounding version you had for the past few weeks - so if you want your non-technical

On Sun, 29 Aug 2004 17:53:29 -0700, Ralph Giles wrote: > On Mon, Aug 30, 2004 at 03:03:28AM +0100, Andy Baxter wrote: > >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample>

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

I'm trying to encode audio tracks as ogg vorbis at very low bandwidths for peer to peer broadcasting. I.e. need to have two streams uploading from modem users, which means the bitrate needs to be around 18-22 kbps. It's not easy to get good quality at this bitrate. The recipe I'm using at the moment is: oggenc Mon-Sep-20-04-0\:3\:27.wav -b 20 -M 22 --resample 17000 --downmix -o

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I want to encode wav file into fixed window length vorbis file. I don't know if "window" is good name for that, but i mean MDCT buffer size. This buffer is passed to mdct_backward() as "in" and it's size is determined by "init->n" variable which is also passed to this function. So, when I say "window

We're trying to dual home samba onto 2 windows domains. We're created 2 separate smb.conf files each having it's own domain specific info. The only differences between them are - different workgroups - different interfaces to bind too - different socket addressees (matches the interface) - and different WINS server IPs. We are using the security = domain model. We've

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with