similar to: Winamp plugin streaming problems

Hello all, I'm glad to announce to you that Vorbis now has full ReplayGain support. If you're not familiar with ReplayGain, take a look at www.replaygain.org. The main features are: a) all songs play back with equal loudness b) removes the need for normalization c) allows for clipping prevention Using it is very simple. Get a compatible decoder (ogg123, XMMS and WinAmp all support it

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

And thats only a player. What about an (Open Source) encoder? Nah, I'll stick with OGG for the time being. Regards, Mark -- Mark de Bokx Internet Service Engineer ICT Communication and Media Services PlanetMediaGroup the Netherlands "They that can give up essential liberty to obtain a little temporary safety deserve neither liberty not safety." --Benjamin Franklin

Shawn sent me a copy of the non-Direct Sound version of this output plugin, and I'm hosting it on my web site's rare software page: http://members.lycos.co.uk/bhafool1/rarities/ I've found it adds far more smoothness and clarity to various low sample-rate audio files, including an 80-minute speech I mentioned previously, due to my soundcard's approach to D-to-A conversion

Hi all, I started testing the RC1 encoder at ftp://sjeng.sourceforge.net/pub/sjeng/oggdrop.exe (based on branch_monty_20010708) On the songs I have tested so far (not much :) I did not hear any stereo issues, but there are some very noticeable problems with the produced files. Songs without much high-end will suddenly have one when encoded. (you'd expect it the other way around) It

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:

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.

Hi all, I have implemented ReplayGain support for Vorbis. If you are not familiar with it, it is basically a method of making sure all your files have equal loudness, remove the need for normalization and prevent clipping during playback. The process is totally lossless, and supporting it requires minimal work. More info about the exact workings can be found on www.replaygain.org (recommended

Hi all, ome people asked if it was possible to make an even higher quality mode by incorporating some of the changes from the first tuned version into the 350kbps mode. I did so and made a new version with this new mode. It gives bitrates from roughly 300-350kbps. That's a lot, but it also gets very hard to find something it artifacts on :) Should be sufficient for archival quality. In

I just wanted to know if anyone has noticed a difference between MP3 and OGG encoded files when having the Winamp equalizer switched on? I seem to be missing bass and the sound doesn't fill the room as much with OGG. With the EQ turned off there is no discernable difference between both LAME and Vorbis encoded files (both at an average of 192 kbps). Is this an error with the OGG Winamp plugin?

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

In case you missed it, RC2 is on slashdot.org too: http://slashdot.org/article.pl?sid=01/08/13/1811241 BTW Gian-Carlo Pascutto wrote on /. that RC3 should appear very soon (in a week or so)? Also, it looks like Wall Street Journal article is not available on-line (at least to non-registered persons). If someone is registered or have a paper edition, please post this article somewhere so that we

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

Hi, Up until now, I've been providing narrowband streams at 24kbps MP3. We've reluctantly been using MP3 instead of say Ogg Vorbis due to wider support. But it's getting to the point where low bitrate MP3 doesn't really cut it anymore. With technologies like AACPlus out there in common usage, people have come to expect better and one can hardly blame them. So I thought

Hi, I frequently encode multi-hour radio recordings in 44.1 kHz 24 bit stereo and am currently using the -6 setting with verify (via the "FLAC frontend" application). Would there be any parameter(s) that may lead to noticeably better compression without increasing encoding time a lot? (My machine isn't the most powerful in the world, even with two CPUs (see sig).) Currently

The winamp plugin would not compile. I am using the Cygnus environment (gcc, and a lot of other GNU utils) on Windows NT. _beginthread() is not found. That doesn't look like a function application code should ever call directly. I converted it to use CreateThread and it compiles and works. Change this: thread_handle = (HANDLE)_beginthread(DecodeThread,0, (void *)(&killDecod eThread)

Hi, I have just integrated libshout2 and libvorbisenc into mserv, my music server (www.mserv.org). It all works well. Thanks for a great product. The problem I have is that when there is a pause between songs of around 10 seconds, Winamp (2.91) decides (due to the 64kbps dropping to a 2kbps stream I guess), that it needs to re-buffer. It fails miserably and generally dumps the connection, or

I have noticed that when playing OGGs in Winamp, the bitrate displayed is always much higher than the average bitrate of the file. This was using the plugin that comes with Winamp v2.80 (v1.2 b7) For example, the latest oggenc encoded a song to 77kbps - I verified that this matched the filesize correctly. Apart from the first and last few seconds of the song, Winamp always displays between 85 -

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

Hello -- I desperately want to start using Ogg compression on music I release on the internet, but I am encountering a fair amount of artifacts. I noticed some in beta4 and was hoping they would disappear with RC1. However, I've been playing with RC2 and I've still been banging my head into a number of problems. A majority of the music I make has long drones with a fair amount of spectral