similar to: OT - RE: When will quality increase be unnoticable?

Richard Polton (Richard.Polton@morganstanley.com) wrote: > Is there a way, in either cdparanoia or ogg vorbis, to encode a live > album which has tracks but no gaps between them? Is the only present > solution to concatenate all the extracted wav files and then to encode > it as a single ogg? You can't cat wav files together. But you can tell cdparanoia to rip the whole CD into

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:

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

On <http://cs.leander.isd.tenet.edu/~mitchell/vorbis_intro.html>: "Each snapshot has a 16-bit number for the "loudness" of the sound, meaning that the scale is fairly fine-grained - it ranges from -32,768 (complete silence during that snapshot) to 32,767 (the loudest volume measurable)." That's not correct. A single sample is not meaningful by itself; each sample

Daniel Déchelotte (maitre_yodan@club-internet.fr) wrote: > Hi Greg, I'm cc'ing the list because I'm sure these questions will be of use to many other people. > Yeah, I also find it hard to resist, but there is still one bit that > concerns me: this huge and not so good-looking black backpack. How big is > it actually? I am suspecting it's huge because I can't

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.

>> Is bit-peeling going to be real (or just a rumor forever)? > Apparently the RC3 streams are capable of being bit peeled, however the > tool to do so was looking likely to be quite complex. I believe the plan > was to have RC4 produce streams that left better hints for the peeling > tool, so as to make the tool simpler and faster, but I doubt we'll see it > until

Hi everyone, My name is Tord Jansson and I'm the creator of BladeEnc. I'm very delighted to see that someone has created an open and patent-free sound format which seems to have the qualifications to be on par (or beat) even the best proprietary formats. Congratulations to everyone involved! I would like to help making Vorbis a popular format for compressed audio and have been thinking

I started thinking about this after doing a little testing with AAC, MP3 and Ogg Vorbis. I was comparing the different formats at similiar bitrates. After a while I finally realized that they all sound more or less the same to me. >From 160 kbps on, I usually cannot detect any difference between a lossy encoding and the original source. If LAME is used, I have to strain to notice anything at

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:

Greetings. In looking through the mail archives, I don't see any mention of parallelism of the vorbis code (particularly the encoder). I was wondering if anyone was looking into this at all? Either with threads, and/or on some kind of parallel machine (e.g., a Beowulf cluster)? I see very little mention of threads in the source code, which leads me to believe that they are only there for

I've put together a first cut for a Win32 DLL using the BladeEnc interface. Right now, it's just a drop-in replacement for BladeEnc.dll that ignores all encoding parameters passed to it and uses the info_A vorbis_info struct (same as the encoder_example). It's not particularly useful as of yet, but you can rename it to BladeEnc.dll and use it with any number of rippers out there

Hi, I released a BladeEnc-style wrapper for the Ogg Vorbis 1.0 codec which works with my CD ripper ACXtractor. You can download it from http://www.marvintec.com/en_download.htm Many thanks for all this great work ;-) <p>Best regards Stefan Schulz/ MarvinTec mailto:support@marvintec.com http://www.marvintec.com http://www.marvintec.de <p>--- >8 ---- List archives:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I notice that FLAC can't handle "broadcast" WAV's in IEEE float format-- the native file format of Ardour and JACK, among other things. It also can't seem to handle 32-bit integer format either. Right now I have to convert WAV's to 24-bit before FLAC'ing it, then when converting it back again I've lost those bits.

Hi all, I've just posted another version of vorb_enc.dll (w/source code) at http://xtractor.sourceforge.net/vorbdll-20000626.zip It will still work as a drop-in replacement for bladeenc.dll (just rename the files that your ripper creates from MP3 to OGG), but will also accept info for the ogg comment header in the vorb struct in the format union of the BE_CONFIG struct. I'll be

I'm trying to get OpenSSH 2.5.2p2 to run on HP-UX 9.05. I've had some decent results, but I'm also seeing some problems. I'm using the EGD (I configured with --with-prngd-socket=/tmp/entropy). * Compiling required some changes, which I've attached. Two of the changes are "hackish", and not at all suited to inclusion in the source tree, but they might point

Thanks Jud, I was able to re-encode my entire FLAC archive into 1.1.4, although on my slow computer (with all other apps closed) it took over 3 days to complete! Only one song returned an error and that was quite easy to fix. I did run into one issue though. I have a FLAC album in which the songs are all 24/96 and REFLAC fails when it runs. I want to re-encode them from compression level 5

On Wed, 6 Jun 2001, Greg Wooledge wrote: :> this is caused by select() being restarted for SIGCHLD on HP-UX due to the :> use of SA_RESTART. can you try the patch below and see if it fixes the :> problem for HP-UX and linux? i've tested on HP-UX 11.11. : :> -#if defined(SA_RESTART) :> +#if 0 : :This doesn't seem to make any difference on HP-UX 10.20. But then, :I'm

I'd like to write a DLL wrapper (BladeEnc/lame_enc style) for libvorbis. I pulled down the CVS tree today, and with very slight changes (missing typedefs for int64_t, int16_t, etc...) got it to build using gcc 2.95.2 in Win95. Unfortunately, the encoder_example segfaults. I get the same results from Cygwin as well. On a linux box with egcs 2.91.66 it builds cleanly, and appears to

One thing I must note before I begin is that, I probably just am not that well skilled at listening, but I can hear no difference between vorbis, mp3, and wav at the bitrates I encoded them at. lame 3.83 (beta) was used to encode the mp3s and mpg123 was used to decode. The current (Jun 8 00) encoder/decoder examples were used for vorbis. mp3s were encoded at 128 kbps, and vorbis is unknown