search for: lossiness

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

Hey all, I recently got a message from my provider about IAX: > We do not recommend the use of IAX. It is a lossy protocol that is > known to cause crackling, loss of audio and other issues. You can > use IAX if you want, but we will not assist with any issues you may > encounter. Does anyone else know about these "known" problems? I'm not sure where this provided got

Curt Sampson wrote: > > > > I've thought of doing lossy compression before on instruments, > but I'd > > > > much rather stick to lossless, at least for now. > > Honestly, stick to lossless. I mean, to the point where you can get > your exact samples back. Sure, an S900 sample is not so great quality, > but having come from the era where I did 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

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

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

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

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

> Steve Lhomme wrote: > > You might try to contact the makers of FruityLoops. Right now their > samples packs are in Ogg format (stored in a WAV IIRC). They like open > and free formats (they also use the LAME encoder). I've thought of doing lossy compression before on instruments, but I'd much rather stick to lossless, at least for now. There are issues to be sorted out

> Good point. File extensions normally represent groups of related > formats. I don't propose differentiating everything (e.g. standalone > FLAC from Ogg FLAC). I do want as a minimum to tell apart these > categories: > > - Lossy audio: Vorbis, Speex. But speech is useful to distinguish > from music, so making Speex separate is not a bad idea. > - Lossless audio:

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

Hello, I have a large collection of audio files contains music in mp3 format, due to need to free space of hard disk, I need to reduce their size. It seems opus is the best format for this purpose, in order to have the quality of original mp3 files, currently I use ffmpeg command to convert them to FLAC and then use opusenc, the official opus encoder, to convert FLAC files to opus. By using one

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

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 instead. > > > >Signed-off-by: Adrian Ambrożewicz

I am interetsed to know views on how does best quality ogg compression compare with SHN or whether this is this really comparing apples with oranges because SHN does not compress that greatly? I am on a Music list that is doing a Tape Tree. I guess Grateful dead have made SHN the cult leader format. -- Raena Lea-Shannon --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg