similar to: lossless ogg encoding with good compression rate

I was just somewhat curious as to what lossless compression is used... is it just huffman? Tony Arcieri --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this

Hi FLAC team. I feel I have found a super high compression way of FLAC. I have tested a 1 hour WAV file of 440HTZ with a 5,25,50,75,100 normalize volume preset. This dramatically changes the compression size of the end FLAC file even though the WAV file size is identical for all 5 WAV files. Only the volume is different. When you renormalize the WAV to its original volume the file is still 100%

Is FLAC a variable bit rate format when streamed? If so, how can it be truly lossless? -- Dennis Brunnenmeyer Director of Engineering CEDAR RIDGE SYSTEMS 15019 Rattlesnake Road Grass Valley, CA 95945-8710 Office: 1 (530) 477-9015 Mobile: 1 (530) 320-9025 eMail: dennisb /at/ chronometrics /dot/ com http://www.chronometrics.com/crs/index.html <http://www.chronometrics.com/crs/index.html>

I'm well aware how compression works. But images and document files do not depend on the relative timing of the data to reproduce themselves. They are in essence only two-dimensional in space, whereas the data in a sound file is time-dependent. The question really has more to do with the decoded FLAC stream output, which I presume is a linear PCM file, e.g. WAV. If FLAC is lossless and

the audio was also 192,000khz sample rate, forgot to mention that, adn here are the audio files, the original, the flac, and the decoded from flac. the archive is 7zip, Idk where to upload it so I'll just send it to depositfiles. http://depositfiles.com/files/90anghniw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, It is possible to make Opus/CELT a lossless coder if I allow a sufficiently high bit rate? We considered using FLAC, but FLAC's latency is well beyond the acceptable range. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20120822/60c69469/attachment.htm

Nicholas is probably right about noise. Another factor would simply be the amplitude of the resulting file. A 275 MB 24-bit file which compresses to 110 MB is probably not very loud. I assume that the average level is somewhat low, with few if any peaks that reach 0 dBFS. FLAC is very good at compressing audio that is not loud. In fact, the quieter the recording, the smaller the

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

Hello everyone, I'm new to this flac thing (started about a week ago) but I have read a lot about flac and replaygain. As far as I understand it, replaygain is lossless in the sense that I can tell my player to ignore the settings or I can even use foobar2000 to remove the tags entirely, hence getting back to the original audio. If that is the case, why is there a warning in the foobar2000

So some research and experimentation I think I found the problem, but I'm not sure how to go about fixing it. If I import one of the wav files into Audacity then export them as "16 bit PCM" then they encode to flac and play fine. The files bitrates are currently at 24. If I export them from audacity as a 24 or 32 (anything higher than 16), flac fails on me. With a bitrate of 24, the

Someone sent me a question late last night and I briefly looked at his file this morning and couldn't figure out the answer, so I'm posting here. A friend has a a ~275MB 24 bit, 48khz stereo wav file of rock music that when compressed using flac level 8 gives a flac file under 110 MB in size. When I dithered his file to 16/48 and converted that file to flac, the resulting flac file was

I found a bug with FLAC v0.6 raw encoding. It appears that the file pointer in the source file is not reset after seeking to the end for checking the size. I've attached a patch. I'm excited about FLAC!! I've been looking for a good GPL lossless RAW audio compressor for use with sound fonts. Sound font files contain 16 bit samples that are word aligned, so just treating it as raw

Brian... You've been both polite and helpful. Thanks. I do understand the dimensional nature of images and sound, though I admittedly glossed over the details while trying to draw attention to time rather than spatial artifacts. What I was looking for was confirmation that a properly designed application would decode FLAC without temporal issues. I believe you've made that perfectly

It seems, that oggenc-ing audiofile, and then flac-ing diffrences between original file and vorbis compressed file gives a little better compression than simply flac-ing. I've tested it on one file only: file.wav 55829468 bytes flac -8 file.wav file.flac 37924329 bytes (0.6793 of original) oggenc file.wav file.ogg 4784799 bytes oggdec -o ogg.wav file.ogg sox tmp.wav ogg-.wav vol

On Dec 2, 2010, at 07:55, scott brown wrote: > My first thought was that the file had low levels (before he sent > me the file), but that's definitely not the case with this file. > There are many peaks that reach 0dBFS. Live, uncompressed music often has peaks that are 4 dB higher, or more, than a typical commercial CD. Such peaks are brief, and would not really affect the

Hello, What is nowadays lossless compression's limit (tested on a large library of music samples, because of course, it depends on the music we do the tests on) ? I often see 53% - 62% of original WAV file's size. ( https://xiph.org/flac/comparison.html ) Are there algorithms that can losslessly compress to, say 40% of original WAV file size (I don't mind about encoding time here) ?

hi flac-dev list! I see, when compressing CD-audio tracks, I can reach up to 60% (ratio = 0.6x) of the original WAV file after compression. I was wondering if the FLAC codec could become as good as reaching 50% of the original WAV file in the future or if we are already at the (almost) maximum compression possible? thx -------------- next part -------------- An HTML attachment was scrubbed...

?hel kenal p?eval (neljap?ev, 14. m?rts 2013 19:02:35) kirjutas Declan Kelly: > On Wed, Mar 13, 2013 at 10:06:51PM -0400, benski at winamp.com wrote: > > Flake is a completely independent codebase. When I used it years ago, I > > remember it being not only better compression but significantly faster as > > well. I believe some of the techniques used in libflake were added to

Hi All, Let me ask this question of the group: When bit rate peeling becomes available, how will the quality of the peeled Vorbis file compare to a file encoded at the target quality directly from the original? So, for: a.wav --> b.ogg (at q6) --> c.ogg (at q2) a.wav --------------------> d.ogg (at q2) how are c.ogg and d.ogg likely to compare in terms of audio