similar to: resampling to 48 kHz

I've had this problem encoding oggs where the output file is small, like 24kbytes for a 4 minute song (tested at 128 and 160kbit). I'm running Windows 2000 and this has happened in oggenc, oggdrop, and CDEX, though I've also been able to get good encodings with each of these. I think the only clean encodings have been .wav's that I've made myself with SoundForge or CoolEdit,

I have yet to speak to a single Mac user who has successfully received a live or archived Ogg stream. I'm been told neither MacAmp nor Quicktime will accept streams, they only play files. Are there any Mac users on this list who can recommend a player (from personal experience) which actually plays Ogg streams? db http://www.subgenius.com/ts/hos.html --- >8 ---- List archives:

I just joined this list, so I'm not sure how many of you are already aware of this. KPFA is a 59,000 watt free speech community radio station in Berkeley, California that has been on the air since 1949. They were the first listener sponsored radio station in North America, paving the way for many other community stations across the U.S. and around the world. KPFA was one of first signals to

Hi Mark Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection? What about the nokia card phone - does it have open source drivers? Cheers Rob On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mark Spencer wrote: > Does anyone (maybe in Europe) know how I could build a GSM compatible > channel for Asterisk, so

Earlier today while listening to CBC radio's national science show "Quirks and Quarks", I was pleasantly surprised to hear that their audio archives are now offered in Ogg, in addition to mp3 and real. This is the first CBC show that I am aware of that offers their archives in formats other than the usual real/wma/qt. For those who don't know, CBC is Canada's only

For those who are interested, it's a weekly radio show produced by the Church of the SubGenius, presenting the bewildering doctrine of PatrioPsychoticAnarchoMaterialism, buried beneath a dizzying media barrage. Hosted by Rev. Ivan Stang, official show archives maintained by yours truly. Now offering 45k and 160k Ogg Vorbis files, listen in headphones: http://www.subgenius.com/ts/hos.html

dd if=$file ibs=1 count=$(($(stat --printf='%s' $file)-2)) of=$file.new of course if you run this on one of the files that doesn't have the extra 2 bytes you're gonna lose something you didn't want to On 11/1/07, Alex Brims <alex.brims@gmail.com> wrote: > Ok, we actually worked this out - there were 2 extra bytes doing nothing at > the end of the files. Opening

Hi, How's the testing going? I compressed 194 individual .wav files (totaling 8.54GB) which contained tracks ripped from many varied albums. I unflacced them and compared their md5 signature with the same from the original .wav. They were all perfect. I didn't use the -V option just in case of any chance of mis-reporting. I hope to test it with the complete collection of ~41GB

> > ( icecast2 only handles .MP3 for fileserving , as far as i know ..) > > Nope. Icecast2 doesn't care about what the format of the files it serves is > for fileserving. (i meant :: .MP3 is handled for fileserving , but .MP3 is not avail for live reencoding, as far as i know .. ) for the record , the OGG streaming w/ icecast2 has been great . tho (seperate issue ) : OGG files

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,

Hi, I'm having problems encoding certain WAV files to FLACs. I'm working for a download store so we get loads of different WAVs from hundreds of suppliers and generally they encode ok, but sometimes we are getting unexplained errors. I've looked through the archives for the last few months but I can't see any discussions that relate to this specific problem. This file:

hello jean-marc and everybody, I keep getting no results when trying to use speex_echo_capture, speex_echo_playback and speex_echo_cancel in a multi-threaded application, as suggested in the manual. Though, the cancellation works properly when i use a file with human voice for far-end input and the same file with echo added in SoundForge for mic input. When i try to insert and remove silence

Hello, I noticed that option --keep-foreign-metadata discards WAV cue markers. Here is how to reproduce the bug: 1) Create a 24-bit 96khz in SoundForge8, add 20 seconds of silence, and add two markers with "m" key shortcut 2) Save it, compress it with "flac --keep-foreign-metadata testmarkers.wav" 3) Decompress it with "flac -d testmarkers.flac" 4) Open the

I'll rearrange a little and respond: --- Mark Powell <M.S.Powell@salford.ac.uk> wrote: > Also, when flac takes input from stdin it fails to > fill in the wav size > fields correctly, whereas shorten has no problems > with this: i.e. >... > You can see it puts a data chunk size of zero in > there. > OK, this has been fixed in CVS. > Flac refuses

Hi!, I wondering if It's possible to change 48khz sample rate limit?, I'm Planing to encode with OGG codec a audio signal but I need that OGG Encoder works with 192khz of sample rate. It's Possible? Any Suggestions?

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

I'm trying to install asterisk@home, I've just downloaded the latest cd from soundforge. I can get it to install ok (network card didn't auto configure - but I worked out how to use 'netconfig'). I worked out how to add a few grandstream budgetone fine. Worked out how to upload music etc. Worked out how to modify FOP. Voicemail and meetme's work fine.

"Didier Dambrin" <didid at skynet.be> wrote: ... > I like FLAC on the paper because of its metadata preservation, in that riff > tag, which is critical for my needs. Try using WavPack, http://www.wavpack.com/ This can losslessly compress 32-bit floating point WAVE-EX files, and faithfully preserves every chunk (which FLAC does not do). It is also free. Regards, Martin --

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting