similar to: Question about volume level recording in Ogg Vorbis

Hello everyone! Please bear with me, I am a complete audio-newbie, and I apologise if I am asking a question that I should know. As most of you all know, DVD's have 5.1 audio, and I want to preserve that in my voribs files. So, to ask the question : Does it support Surround sound? Thank you very much!

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

Hello folks, Although Jack graciously permitted me to keep tinkering on the Postfish alongside OggFile work ;-) allowing it to sit there in a partially tinkered state continued to be a distraction. So yesterday and today I cleaned up the source repository and put together a prerelease. With this prerelease, I'm putting the project out of mind until I put out a beta of OggFile. Actually,

I'd like someone to clear up, or point me to some document clearing up, the character encoding for tags. In some of my OGG:s, non-ASCII characters show up like this: ? -> a ? -> "a ? -> "o etc. While when I set new tags, they later all show up as '#'. What kind of encodings are used in these cases. Which is preferred? (I have no LC_* environment variables set,

I've been tinkering with vgplay (my fp gain function and replaygain support code) as well as looking at FLAC replaygain code, with an eye to combining them into a general gain effect plugin helper, however I noticed that the November 19, 2003 Ogg Traffic talks about postfish, being a sound postprocessor with a noise filter and 'a revolutionary declipping filter', that Monty is

Hi, I bought an iRiver HP-120 which can read ogg files (via a hard drive). Many people experiment that reading ogg file decrease the baterry charge more quickly than with mp3 at the same bitrate. The only raisonnable reason of that mysterious thing is that reading ogg file is not sequential onto hard drive, and the hard drive head must move more than for reading an mp3 at the same quality, and

Hello! I'm running samba 3.0.20 on Solaris as PDC. But I cannot add Windows Server 2003 clients to the domain. After authenticating as root, I get the following error message: "The parameter is incorrect." in Windows. The result is that the computer isn't added to the domain. :( Any ideas? Thanks. The samba log says: [2005/12/05 15:54:37, 2]

Hi, I 'm happy when I see that version 1.1.1 has been released, but disapointed too by the fact that the setup has not been updated onto the link HYPERLINK "http://www.vorbis.com/download_win_1.0.1.psp"http://www.vorbis.com/down load_win_1.0.1.psp, which can only allow to download version 1.0.1. Could you tell me where I can find the complete list of fixes / improvement of the

I just got myself a portable Ogg player (Jens of Sweden MP-120). It turns out that it doesn't like to play Oggs I've encoded with the 1.0-rc versions of oggenc. Is it in any way possible to re-encode those with as a post-1.0 Ogg Vorbis file? (Without first decompressing them, of course.)

Hi: If you can record your mic but it's not working in XMMS, this suggests an issue on the XMMS side. Are you invoking the XMMS Recorder plugin as detailed in the Readme? Geoff. <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org'

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Hi everybody: Here is a new Ogg Traffic, with release announcements for Vorbis 1.0.1, Speex 1.0.3, and Icecast 2 Beta 1. The HTML version is at http://www.vorbis.com/ot/20031119.html. Enjoy! -Carsten <p>Ogg Traffic for Wednesday, November 19, 2003 Author: Carsten "Purple" Haese Contact: [1]carsten@xiph.org Date: November 19, 2003 Table of Contents 1

I?m pretty new to * but it?s got my vote so far. I?m trying to test calls coming into my system but * won?t do anything useful with the call. There are no errors when starting * but the following is received when I try to call the machine over a channel of a T1: *CLI> -- Starting simple switch on 'Zap/1-1' -- Starting simple switch on 'Zap/2-1' -- Hungup

When trying to compile samab-2.2.0 on a Ultrix 4.5 box i get the following error: . . . . Compiling smbd/blocking.c Compiling smbd/sec_ctx.c smbd/sec_ctx.c: In function `initialise_groups': smbd/sec_ctx.c:227: warning: comparison is always false due to limited range of data type smbd/sec_ctx.c:227: warning: comparison is always false due to limited range of data type Compiling smbd/vfs.c

Hi All, I have what I think is a ZFS related bug. Unfortunately my simplest test case is a bit cumbersome and I haven't definitively proven that the problem is ZFS related. I'm hoping for some feedback on how to move forward. Quick background: I rip my CD's using grip and produce flac files. I tag the music using Musicbrainz' Picard and transcode it to mp3's within Picard

hello All, i have one issue with Asterisk Meetme Application i am recording through Meetme channels through option *'r'* and format for recording a file is '*wav*' lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. i have very strange problem of meetme_recording , once conference starts recording file having a *recording is 2x faster *than normal recording .

Hi gang! I'm trying to record (to .wav files) sound coming in on the line input of my sound card (in this case, output of a tape deck). For LISTENING purposes, I can control the level with the Gnome Mixer. However, the RECORDING level is not affected by the mixer. As a result, the recorded audio is badly clipped and horribly distorted. I don't understand enough about how this stuff

Hi all Does anybody know where this command is supposed to go? Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) In the queues.conf file it talks about it. So I naturally thought after I set up my monitor with monitor-format = wav monitor-type = MixMonitor That I could put a lame command in there to convert and move the file elsewhere for backup with

I haven't seen much activity on the list concerning Digium's IAXy standalone FXS device. Are they worth $100 ? Do they integrate well with * as an extension's FXS ?

Hey all, There are a couple posts about the possibility of using Red 5 to record audio through a user''s mic to the server, through RoR apps. I wonder: besides Odeo (I think they''ve developed an in-house app?) are there any examples and/or demos of this being done? Or, are there any reasonable java sources that actually *do* work to take mic input from a user''s machine,