search for: trebles

> Date: Fri, 05 Dec 2003 13:22:53 -0500 > From: Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> > > I think I see what you mean, though I haven't been able to listen to > your wma file (not everyone has a wma decoder). The problem probably > only lies in the VBR tuning for wideband which hasn't received much work > yet. One way to check that is to encode in

Hi, I have a problem where some PRI channels get stuck in a "Call" mode. If I do a zap show channel XX, it shows as "PRI Flags: Call". However there is no calls on that channel. Trying to force a hangup does not work: [root@neil1 Dialer]# asterisk -r -x "soft hangup zap/27-1" -- Remote UNIX connection zap/27-1 is not a known channel Any ideas?

I tested vorbis. I sounds good, but when I play something in mono, trebles are much stronger. Perhaps is stereo phase of hi bands removed to improve compression ? I't a good idea (it's not audible in most music), but it should also provide an option to preserve them. Is there docs that explains how dows vorbis compress sound ? oggenc (yearstday's cvs) creat...

[Apologies if this gets through twice. I sent it first without subscribing, but it seems like it got stuck in the moderation queue, so I subscribed and re-sent it.] I'm doing some work on audio fingerprinting for a school project (more precisely, my master's thesis. I got a hint on #vorbis that I might want to look into the internal floor representations in libvorbisenc to get out audio

I've learned that I can make a tag with vorbisgain that tells the player at what volume a track is to be played to achieve a certain level of loudness. Is it possible that there will be a tag that tells the player for instance to add more bass when it plays a track? Ore more treble? With such tags it would be possible to customize the sound of a track according to my wishes - without

thanks for getting back to me, i have uploaded a zip file containing some sound files that demonstrates the issue. http://www.bogus.net/~olav/ess.zip this contains s.mp3 original wav file (mono) converted to top-quality mp3 (370K) s.wma windows media encoder with 19khz voice compression ( 62K) s-2.spx speexenc --vbr --quality 2 on the wav file ( 63K) s-9.spx

Hi there, I'm new to this list, and in fact rather new to the Ogg/Vorbis codec altogether. I recently downloaded the test files from the test conducted by the German computer magazine c't (this was mentioned by someone here recently). The idea was that you were given seven WAV files each for 64kbps and 128kbps, and had to rate them. The files were randomly chosen for each participant.

I have found the following opinion on Vorbis' talk page at wikipedia. It might be something interesting to consider, I suppose. ~*~ "Quality settings run from -1 to 10 and are an arbitrary metric; files encoded at -q5, for example, should have the same quality of sound in all versions of the encoder, but newer versions should be able to achieve that quality with a lower

Hi Guys, I have a problem with IE6. Using any version of prototype and any version of scriptaculous IE loses the ability to select text by dragging the text cursor over it. You can double/treble click text, or use shift and the arrow keys but you cannot use the mouse. This applies to text in both input and fixed elements. Has anyone else got this, or knows how to fix it? Cheers Paul Shannon

OK, I've organized this. It's not possible to average 32kb/s stereo with Vorbis using VBR. Quality -2 stereo proeduces roughly 32 to 38kb/s which is too high for many modems. I would have to use ABR, but last I tried that, it utilized too much CPU for my 950Mhz PC, and oddcast doesn't appear to support that option anymore. I've tried AACPlus 32kb/s mono and there is very

I really don't know if this is the same problem that was reported by Ingo Saitz (I really couldn't say which one was 128kbit and which was the original wav when blind-testing. The original had some distortion that perhaps does somehting with my cheapo soundcard) but here's a description of what i've found: When doing some sample encoding with rc2 (rebuilt rpm with latest redhat

I would very much like a lowpass option because for FM radio broadcasting I don't want to encode frequencies above 15khz. I'm waiting for this option before switching to ogg from mp3(lame). Ross. > -----Original Message----- > From: owner-vorbis@xiph.org [mailto:owner-vorbis@xiph.org]On Behalf Of > Gian-Carlo Pascutto > Sent: Tuesday, 4 September 2001 01:46 > To:

> i have done further testing, and even at constant bitrates, wma is far > superior, from an "overall listening experience" point of view. OK, one thing at a time, so will you please leave wma out of that for now? > it seems that: > > speex maintains the crispness/treble of the recording, but with the > cost of computer-ish background noise, like turning into

Hi Christopher > I was under the impression that "PCM equalizer" means it only equalizes the > PCM codec, i.e. WAV files. No, not at all. It equalizes the audio after the decoding of compressed audio. > Have you noticed a significant quality > improvement in the EQ for compressed media: MP3, Vorbis, MPC? Yes, definately. The Nullsoft MPEG Input plugin has it's own

I should have provided a link to my low bitrate vorbis stream. It's a mostly easy listening world music format: http://soulfm.stationplaylist.com/listen.m3u Regards, Ross.

I've been searcing this newsgroup and other places for a solution to this and finally decided to post and see what anyone has to say. I am running Ubuntu 6.06 with Wine 0.9.30. The software that is giving me problems is Finale 97, a music notation program. Finale installs Truetype (.ttf) fonts for the special music notation characters. The display on the screen is fine. When I print, the

Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I've seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 +

I have successfully installed and launched Sibelius (music typesetting software) on linux (with codeweavers preview 4, w/ a fake_windows install). It works fairly well, and would probably deserve a 4 on the app database. Except for the fonts, which kind of defies the purpose of a typesetting software! Here are the symptoms and some log activities : * No musical symbols are displayed (except

Just a (I hope) quick question: Are there any advantages (related to sound quality and file size, not to encode/decode speed or segfaults) of using 1.0 instead of RC3? I as it because I'm a Debian user, and I track the stable branch, which means I'm stuck with the vorbis tools version RC3, unless I want to use unofficial apt sources (don't like it that much), or compile my own