similar to: This works: low bitrate encoding

Hey guys. I think Vorbis is pretty cool, but since the current OggEnc only offers 44.1kHz, it limits what I wanted to use it for. So I've been using Lame to get 16kHz mono Vorbis files. I'm curious about whether Lame does Vorbis encoding the "right" way for non-44.1k stuff, or whether it just encodes as it would for 44.1k & changes the sample rate on the output, but I'm

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

I am very excited to see the Vorbis 1.0 out. Unfortunately, neither Sonique 1.808a nor Winamp 2.5 is willing to play the resulting files. Only silence. I tried the simple case of 'oggenc track1.wav' and it created the expected track1.ogg, 3.5MB in size with avg. bitrate of 110kbps at default quality 3. These applications worked fine with a much older version of oggenc. Maybe Beta 2

Hi there, I have a question on ogglame, especially on its compression quality. I've download the binary package of ogglame from www.vorbis.com. It works fine. I've also tried LAME 3.85 beta that supports the encoding and decoding of Vorbis streams. I believe that ogglame is a beta version of LAME. I built LAME using libvorbis in the CVS nightly snapshot fetched on July 19th. I fixed

Hi. Joined the list since this seems to be the only point of communication. Just have two comments/questions. 1) I looked all around the developer's site and vorbis.com and don't see any mention of where to send bug reports for 1.0 beta. I'd think it would be a FAQ. 2) My bug for the new beta. (I'm hoping someone of the PTB is reading ... I definitely don't belong on the

Hi. Joined the list since this seems to be the only point of communication. Just have two comments/questions. 1) I looked all around the developer's site and vorbis.com and don't see any mention of where to send bug reports for 1.0 beta. I'd think it would be a FAQ. 2) My bug for the new beta. (I'm hoping someone of the PTB is reading ... I definitely don't belong on the

Don't know if anybody is still missing the lame oggenc features for resampling, lowpass/highpass filters etc, but I wrote a little script that uses sox to do all the stuff I need to real-time encode oggenc from the radio, or any input device. #!/bin/bash DATE=`date '+%m-%d-%Y-(%H.%M)'` DESTIN=/video/music/perftoday export DATE=$DATE'-PerformanceToday.ogg' sox -V -r 44100 -c

I'm now trying to use an m3u formatted list to pass to ezstream rather than stdin. Since all of my audio files are in .wav format, and I have oggenc, I see no reason to decode them before ogg encoding them. Is there a configuration I can use to make this happen? I tried omitting the decode parameter and ezstream complained. Here is what I have now: <ezstream>

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,

------- Forwarded Message From: Jesse Hammons <jessehammons@yahoo.com> Reply-To: jhammons@alumni.princeton.edu Subject: VBR encoding in ogglame? To: vorbis@xiph.org Cc: vorbis-dev@xiph.org MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Hello, I ran a little test where I encoded an audio file at a bunch of different data rates. The results are here:

------- Forwarded Message From: Jesse Hammons <jessehammons@yahoo.com> Reply-To: jhammons@alumni.princeton.edu Subject: VBR encoding in ogglame? To: vorbis@xiph.org Cc: vorbis-dev@xiph.org MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Hello, I ran a little test where I encoded an audio file at a bunch of different data rates. The results are here:

Hi. Sorry if this is the wrong list - a pointer to the right would be appreciated. I have a number of Ogg files, that have a VBR bitrate of approx. 256 kbps. I have bought a portabel Ogg player (iRiver iFP 795) that unfortunately only support bitrates up to approx. 225 kbps. So my question is, if there is a tool to change the bitrate down to e.g. 220 kbps. I know I can do this with sox, or

Hey all...I'm trying to use the decoder_example.c in the examples directory to decode a .ogg file that was encoded by using the binary encoder ogglame.exe and encoded via the encoder_example.c. I'm running into a problem with the decoder though. It plays FINE under the WinAmp plugin but I when I try and use the decoder_example.c to decode the file it says "End of file before finding

I am trying to convert some a few WAV files to ogg format (using vorbis-tools-1.0beta4 oggenc). I tried: $ oggenc -o nash.ogg nash.wav ERROR: Wav file is unsupported subformat (must be 16 bit PCM or floating point PCM ERROR: Input file "nash.wav" is not a supported format And: splay -d - nash.wav | oggenc -r -o nash.ogg - The resulting file when played with ogg123 played so

Okay, this is similar to how the mp3 streaming was. The question is then, what is the MIME type for the playlist itself? For mp3s, it used to be audio/x-mpegurl?? Thanks, Matt >You need to create a playlist containing the URL of the Vorbis file. Then >the browser will just download the playlist and hand it off to your player. >Which player are you using? Not all of them have >HTTP

Hi! I'm very impressed with the progress Ogg Vorbis has made, ever since getting that T-shirt at LinuxWorld Expo :-) Using oggenc, I've run into a problem, though: "Mode initialisation failed: invalid parameters for bitrate" This happens when 2 conditions are true: 1) the input file has a sampling rate less than 44100, and 2) the -M parameter is used to specify a maximum

Hi all, I'm very new to audio streaming, so please excuse my ignorance. I'm using icecast2 server and muse source client on gentoo linux. Some mp3 files that I'm streaming exhibit severe static noise, in fact, the noise is present always but is most noticeable for the records that are currently playing, http://24.57.213.147:8000/aha.m3u (I'll keep this alive for a while) I

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

I recently bought a DVD Audio disc and I searched the web and wasn't really able to find much about DVD Audio and Linux. I have a few questions that I was hoping some people on this list might be able to answer. How can I extract the uncompressed DVD Audio (not the DTS Audio portion of the disc, but the 96khz/24bit/5.1 pcm part)?

Hi, Was wondering if any gurus could tell me if it is possible to stream Ogg on very low bandwidth links (for example satellite phone) where the expected bandwidth would almost certainly be less than 19Kbps, and probably more like 10Kbps. I am not after great audio at this bitrate, mono with an audio bandwidth less than 5KHz. I have done some testing (not streaming though) listening to 8Kbps MP3