similar to: Low bitrate: when?

Back in 2001 Patrick Godeau posted to this list a link to an awesome track called stonehenge. A bunch of us here downloaded it. Somehow I managed to hose my copy and was wondering if anyone on this list still has it around and could post a link or email it to me? I absolutely love that track and I can't even see where I could purchase it... as far as I know it's a one of a kind gem,

Hi there, I have developed pure Java Ogg Vorbis decoder, JOrbis. By the recent improvements on JOribs, its 'simple' player, JOrbisPlayer, will enable you to enjoy low-bitrate/mono streams on the web browser with J2SE 1.3(or later)Plug-in; Java plug-in; I have provided digitally signed JOrbisPlayer at http://www.jcraft.com/jorbis/player/JOrbisPlayer.php # This signed applet is signed

I am writing a long-run sound recording application using OggVorbis. The user can adjust sound quality parameter to make balance between storage space and sound quality. The input sound is 22050Hz, mono. To maximize the recording capability on a given storage volume, we want the result bitrate as low as possible. I use -q0 parameter and it produces a 32.0kbps stream. It is too high (our

dear vorbis developers, it would be very useful, if oggenc would support constant low-bandwith bitrates for testing. something from 20kbps to 44kbps, so anyone could test streaming over modem and isdn. audio quality doesn't matter. mörk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

At SxSW I ran into the Xiph gang and someone said something in passing that the new version of Ogg Vorbis would have much-improved low-bitrate quality. Wondering - true? Details? ETA? At CD Baby all lo-fi soundclips are still in the SFERA format*. At 20k bitrates, they sound GREAT which is why I stick with it, but I'd much rather switch them ALL over to OGG if there was a way to make

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

Hi all, I'm running an Icecast-2.2 server with at peak times some 50 sources and 500 concurrent listeners all using low-bitrate 16kpbs streams. I'm experiencing some connection losses at these peak times ("Client connection died" message in error.log). The machine running Icecast has a 100Mbit connection to the internet. It is a Celeron 2.4Ghz machine with 1Gbyte of main

Hello- I recently downloaded Verbs and started playing with the encoder. The quality was great when I encoded a few samples at medium and high bitrates, but when trying to encode the same clips at 16 kbps, artifacts are audible during playback- short bursts of screeching noise, not just distortion. I'm using oggenc 1.0 beta 4 to encode, sonique to play back on Windows NT. I've also

Hi list, I'm doing some experiments with relatively low bitrate encoding, and can't find a player/decoder that works consistently. Winamp and freeamp seem to choke up when you try to play low-bitrate vorbis files. There's mention of an ogg123 player/decoder, but it doesn't seem to be includeed with the vorbis-tools package as advertised. (Does anyone know where it is? Is

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

Hi, is there any means to (simply) change the (nominal) bitrate in the encoder to (much) lower rates ? dev --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

Hi all, I have a question about the relationship between the "queue-size" and the "burst-size" <limits> parameters. Q: When a client connects is the burst-size immediately added to the queue-size? If this is the case then the slack for a 16kpbs client goes down from 50 sec (102400[queue-size] / 16kbps = 50 sec) to 18 seconds ( (102400 - 65536[burst-size]) / 16kpbs =

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

I am totally new to icecast, and would appreciate a pointer in the right direction. My application is to encode audio (at the linux server) at a low bit rate, say ~16 kbps, for transmission to a decoder at another location. The audio source will be bandlimited (basically voice grade circuit such as you'd find in a telephone application.) The decoder end (connected via a TCP/IP session)

Hello, the voice-communication TeamSpeak (www.teamspeak.org) is currently testing a version that supports speex codecs. The quality for high bitrates is quite good. BUT, the low-bandwidth speex codecs that are currently used arent very good. What I did to find this out: I comprared a speex AVB with 6.3 KBit/sec (total, overhead for packets and stuff included) and the 6.3 Kbit/sec Celp Codec

Am Dienstag, 13. Mai 2003 03:22 schrieb Allen Drennan: > Hello, > > HawkVoice doesn't have a 6.3kbps codec for CELP, it has a 4.5kbps CELP > codec and I do not believe it is being used by TeamSpeak. The 6.4kbps CELP > being used in TeamSpeak, to which you are referring I believe comes from > Lernout & Hauspie's LHACM.ACM file which it appears you are redistributing

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

Hello, good morning ! Running Icecast2 and Ices2 with live audio from Line IN, what is the correct way to change the bitrate ?. In the Icecast2 configuration file I set: <mount type="normal"> ????<mount-name>/entzun.opus</mount-name> ????<bitrate>128</bitrate> </mount> But it seems that is not enough since the bitrate does not change. Receiving the

> From: rillian (rillian@telus.net) > Date: Mon 29 Oct 2001 - 22:21:51 PST > > On Mon, 29 Oct 2001 dongoodman wrote: > > > >> frankly, i'm rather tired of the lack of free ogg vorbis players > for Mac > > >> OS; it doesn't look like iTunes is going to support the format > anytime > > >> soon (and i have some real issues with its