similar to: [Q] Lossy compression background information book?

Hello everyone, I'm a student at the Universtiy of Delaware, and will be soon starting some research on the effects of lossy audio compression on speech sounds. I will be preforming test with both mp3 and vorbis. First of all, if I use the '--ogg' switch to lame, does lame use GPSYCHO to encode the wave, or some other psychoacoustic model (perhaps one designed for

Hello, in the background brief documentation papers about the mathematical background of vorbis are mentioned. Where can I found this papers or some other information about the used compression algorithm. Sascha --- >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

Situation:? * remote virtual server with very little storage (estimate: I can spare about 40G for music) * local music collection of ~80G in all sorts of formats - lossy in varying quality, some lossless too Vision: * stream my whole music collection randomized so I can listen to it anywhere Plan/Idea: * Locally transcode everything to one format that results in files that are?

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

>> ..But sadly none of FLAC, WavPack or OptimFrog could compress the >> pre-processed song better, or hardly. And considering you'd also >> have to add >> the pool of frames, it would end up worse. > > This surprises me. Have you tried aligning your frames to the > standard FLAC frame size? > Not at all, because I have no idea how it works internally,

On Aug 8, 2009, at 23:11, Didier Dambrin wrote: > Electronic music quite often doesn't leave a computer these days. > And it > mainly consists of drums, synths & vocals/effects. Drums are often > samples > sequenced at sample (not sub-sample) accuracy, thus repeated (of > course if > the song was post-resampled, there will be sub-sample times). Good point. I

compression : xz, Level 6 here some compression rates I experienced : Diskspace usage : user1: uncompressed : 2.3GB --> compressed 1 GB user2: uncompressed : 6.2GB --> compressed 3.9 GB just for your reference Ing. Robert Nowotny Rotek GmbH Vienna/Austria

Hi Everyone, My first posting to the list. Longtime Ogg fan, have created many .oggs, but this is the first time needing help. I've been transferring CDs to Ogg files using crip. I've been in the process of setting up a bunch of mules to transfer my music collection to electronic format, but I've found an oddity: >bash$ oggenc -q 5 oddity.wav >Opening with wav

From a120342e0d3d20962396e6bf5bd5ac30c66b5983 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Adrian=20Ambro=C5=BCewicz?= <adrian.ambrozewicz@linux.intel.com> Date: Tue, 7 Jan 2020 11:07:08 +0100 Subject: [PATCH 1/1] Fix lossy conversion of Content-Length MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Actual variable holding content length is int64_t,

On Sat, Jan 08, 2011 at 12:54:01AM +0100, jorgen at anion.no wrote: > > I think we agree now on that the "find mp3 before encoding" feature would not be a good idea to implement in the flac core. As Brian pointed out, it might be a better idea to create a program that automatically checks if a flac might have been an mp3 source. It would be more versatile to check if the

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...

On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: > Actual variable holding content length is int64_t, but it was assigned > by explicit cast to size_t. On 32-bit systems it's a lossy conversion, > so it was replaced by casting to int64_t instead. > > Signed-off-by: Adrian Ambrożewicz <adrian.ambrozewicz@linux.intel.com> > --- >  plugins/curl/curl.c | 2 +- >  1 file

W dniu 1/9/2020 o 14:07, Richard W.M. Jones pisze: > On Wed, Jan 08, 2020 at 07:19:56AM -0600, Eric Blake wrote: >> On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: >>> Actual variable holding content length is int64_t, but it was assigned >>> by explicit cast to size_t. On 32-bit systems it's a lossy conversion, >>> so it was replaced by casting to int64_t

Any comments? > http://slashdot.org/article.pl?sid=01/07/13/1558239&mode=nocomment > > Lossy Music Formats Compared > Posted by michael on Friday July 13, @12:45PM > from the is-it-real-or-mp3 dept. > > Nicholas writes: "Today's Washington Post has an article detailing > the results of having a "a diverse panel of listeners: two members > of the

On Thursday, 9 January 2020 17:41:58 CET Adrian Ambrożewicz wrote: > W dniu 1/9/2020 o 14:07, Richard W.M. Jones pisze: > > On Wed, Jan 08, 2020 at 07:19:56AM -0600, Eric Blake wrote: > >> On 1/7/20 4:13 AM, Adrian Ambrożewicz wrote: > >>> Actual variable holding content length is int64_t, but it was assigned > >>> by explicit cast to size_t. On 32-bit

Thanks - I have pushed this now. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones Read my programming and virtualization blog: http://rwmj.wordpress.com virt-df lists disk usage of guests without needing to install any software inside the virtual machine. Supports Linux and Windows. http://people.redhat.com/~rjones/virt-df/

Brian Willoughby <brianw at sounds.wa.com> wrote: > On Jan 7, 2011, at 16:27, Declan Kelly wrote: ... >> As the human hearing can't really tell direction with lower >> frequencies, >> it's not as essential. This same shortcut is why most movie "surround >> sound" systems have only one sub bass channel. > > In this case, you have been misled

Actual variable holding content length is int64_t, but it was assigned by explicit cast to size_t. On 32-bit systems it's a lossy conversion, so it was replaced by casting to int64_t instead. Signed-off-by: Adrian Ambro?ewicz <adrian.ambrozewicz at linux.intel.com> --- plugins/curl/curl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/plugins/curl/curl.c