similar to: quality question?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 If I'm going from .wav --> .ogg -q9 --> .ogg -q7 do I suffer any ill effects or is it just like going from .wav --> .ogg -q7 ?? I ask because I think I want to go down a bit more as the size of the files are around 10mb or so. I can get them from source again and go straight to - -q7 if need be. - -- Robert Cole -----BEGIN PGP

Hi all, I used vorbis several years ago because of its freeness and VBR, but after playing ha.org and getting an mp3-capable discman, I switched to APS and API mp3 encoding. Now, I am liking musepack a lot, but the flexibility is more "there" for the vorbis format, including players, tagging comments (lyrics in the comments - killer idea!), etc.... However, I am a bit worried

Can I reduce the bitrate of an OGG file without recompressing everything? It would be useful to convert high bitrate ogg to lower bitrate ogg for portable players. Olaf <p><olaf@ kjws.com> for every kind of mail, except spam! :-) --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message

Hello. First of all, thanx for that great codec. I would encode all my music with Ogg Vorbis in the future, if it hadn't been for some samples I encoded for test purposes: Everything was great as long a I used "natural" sounds/music, but certain pieces of synthesized music (specially techno) contained very audible artifacts even at high bitrates. So I just went on using mp3 for

Hi I made a CD-backup with cdparanoia | oggenc and I tried to use different values of -q listening to the effect by cdparanoia | oggenc -o - | mplayer -. The best quality was 8 -- why? Oggenc version was the precompiled 1.0 from Mandrake 9. --- >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

Hi, I have this idea: It can be useful to have for each frame two variables stored in bitstream that will hold information about starting and ending position of \"valid\" samples in the frame. With these additional information it will be possible to do simple editing (like cutting or concatenating two streams) without decompressing/recompressing at sample-granularity. Simple example

Git version of the FLAC encoder prints error message: ERROR: Compression failed (ratio 1.xyz, should be < 1.0). Please contact the developers. in the following cases: a) recompressing from FLAC 1.2.1 (sometimes) b) encoding very short wav files (around 5k samples) c) encoding white noise test signal.

R-developers, I have a binary R package built using R 2.14.1 that I would like to run on R 3.0.0. Unfortunately, the original source code is unavailable, so I cannot rebuild the package as R 3.0.0 requires. Is there a straight forward way of converting the package (.rdb, .rdx and .rds files) in the binary package from a 2.14.1 version to a 3.0.0 version without the source code (perhaps

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

If I use flac with piped input it warns: FAILURE: Compression failed (ratio 495232,432, should be < 1.0). This happens for some files for one or more of the following reasons: * Recompressing an existing FLAC from a higher to a lower compression setting. * Insufficient input data (eg, very short files, < 10000 frames). * The audio data is not compressable (eg a full range white noise

Suppose I have a particular version of a largish compressed archive, most likely a .tgz or .tbz2, and that a remote machine has a newer, and only slightly different, version of the same archive, where most of the content hasn't actually changed much. I might attempt to obtain a copy of the newer archive by first copying my local older copy to the newer name as a file to update from. My

I just setup a new SSD with my laptop root filesystem, and at the time I though, "eh, I''ll just use zlib compression during the first copy, and then switch to lzo afterwards to maintain write speed when I''m using the laptop after the copy and reboot". Now, I rebooted with the new ssd and zlib compressed rootfs, and it seemed to boot slower than it did before with the

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

A large test I ran on flac 1.0 recently finished so I thought I'd post the results. I took about 60 CDs, totalling around 30 gigs uncompressed, and compressed them all using all 10 of flac's default compression modes (-0 through -9). The CDs are of a wide variety of music; I think the only major genres not represented are country and rap (freudian slip). Anyway, the raw numbers: Opt

(I've already posted this message month ago, but nobody answered, may be it was not delivered to newsgroup?) Hello, All. I wrote an encoder-decoder based on example from OGG-Vorbis SDK. This encoder can encode a large amount of small WAV-files with equal parameters into one sound archive. Then I decode this sound archive back into large amount of small WAV-files. Theese files are

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all: I have a question which, maybe, could go to the Theora FAQ. How can I join two theora video files in one without recompressing them? Can I just join them one after the other, like with Ogg/Vorbis? I've created a front-end for Thusnelda and want to offer extra features not yet in other software. My idea is to do multithread compression

On 14-03-13 20:02, Declan Kelly wrote: > The next official release of the FLAC command line should really have > a "-9" option for absolute maxed-out big-memory CPU-burning compression. No. If you want such things, try TAK, OptimFROG, Monkey's Audio or even LA, you'll lose hardware compatibility anyway and they do much better than FLAC will with a -9 option. FLAC 1.0

----- Original Message ----- From: "Silvano" <silvano at uel.br> To: <r-help at r-project.org> Sent: Thursday, June 30, 2011 9:07 AM Subject: Tables and merge > Hi, > > I have 21 files which is common variable CODE. > Each file refers to a question. > > I would like to join the 21 files into one, to construct > tables for each question by CODE. >

<PRAISE> First, thanks to everyone for their contributions to flac. It is obviously a significant piece of work with very good performance. For me, the open-source and anti-copy-protection principals of the project make it the best choice among lossless compressors even if there are closed-source projects with marginally better compression. I firmly believe that the open-source process

I just read some of the discussion on the list about 'bitrate peeling' and remembered an interview of Monty that I have read recently. In it he says that Vorbis uses MCDTs <sp> and that these are theoretically reversable. And now, I learn that theoretically we can use bitrate peeling to make smaller files from larger ones, and that leads to my question. Could I theoretically