similar to: audio analysis tool?

Hello. Me again. Have you tried to encode piano solo? Noticed high bitrate Opus gave? And there's also artefact at 15kHz which wasn't in the original audio. Visible with Spek program. Download FLAC and Opus both files, new link: http://www.filedropper.com/example_3 FLAC full: 1084 kbps; FLAC solo: 465 kbps. with --bitrate 160: Opus full: 158 kbps; Opus solo: 190 kbps. Included also Spek

When encoding a wav file with vorbis 1.0, I notice the following message: [eds@crash eds]$ oggenc -q 5 grind1.wav Skipping chunk of type "cue ", length 52 Opening with wav module: WAV file reader Encoding "grind1.wav" to "grind1.ogg" What's this "skipping chunk" all about? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg

Hi all! I would like to use Ogg-Vorbis to encode a non audio waveform. My waveform is in .wav format, on 16 bit mono, with frequency range from 100Hz to 100MHz. It's about 100MB lenght. I need to compact it with lossy for net transfer. Is there something like this, already done, that can help me ?? How can I measure the distortion that Vorbis introduce? I'm sorry for my bad english.

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

I've been installing OpenSSH 2.9p2 onto several RedHat Linux machines, after compiling in the GSSAPI/Kerberos5 patch from here: http://www.sxw.org.uk/computing/patches/openssh.html I've been using ssh both to let users in via passwords and Kerberos tickets, and both have been working fine... except for one irritating machine, which (for no good reason I can see) fails when using kerberos

Hi All, What is the status of the oggpcm project? I'm investigation solutions to the following problem: losslessly encode double-precision mutli-channel timeseries data in a format that is compatible with free (libre) internet streaming technologies and that permits diverse metadata to be encoded with the stream. flac isn't suitable because it only supports integer data, lossy

No offense, Segher, but the output quality of this thing is awful. =) I'll disregard the fact that, at least with *my* compiler, the source tarball I downloaded reduces every packet to zero bytes, which isn't terribly interesting. I decided to set the byte reduction to something constant: I started by dividing each packet's size by 2 just to see what would happen. The resulting ogg

Hi, I am looking for a way to stream live audio using the FLAC audio format. After some analysis myself I found various lossy alternatives, also in higher levels of kbps (like MP3 320 kbps). But I do want to challenge and be sure whether FLAC might be a possibility. So I am specifically looking for an encoder and server solution supporting live streaming FLAC audio. Do you have a

Apologies; a filter mistriggered on the original send of this message. Monty Date: Fri, 10 Jan 2003 01:11:16 -0500 (EST) Message-Id: <200301100611.BAA12395@biohazard-cafe.mit.edu> To: vorbis@xiph.org From: ben-extra@MIT.EDU Subject: Hi, I am hoping for some guidance. I have a *bunch* of audio in the form of realaudio streams. As is, they are of a quality similar to fair AM quality.

Why does opus scale down frequency from 22k to 20k ? 64kbps<https://lh5.googleusercontent.com/-DGsEd1ijTBs/UDE3OFVrE5I/AAAAAAAADBo/bFcWQ6I4-Zk/s588/opus_64_test.png> 128kbps<https://lh3.googleusercontent.com/-3Wfvq6OPGRw/UDE3OClxEdI/AAAAAAAADBk/vsBJ9Gcd_pU/s589/opus_test.png>

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

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

Hi all, [For the uninitiated: a "peeler" is a program that transforms a Vorbis stream into a smaller, (somewhat) lower quality Vorbis stream, and does so quickly, by just throwing out some data.] After having prototyped several peelers that aim to peel to a certain filesize, or to a certain quality, with mixed success, I've now taken a different route: a peeler that aims for the

Hi all, [For the uninitiated: a "peeler" is a program that transforms a Vorbis stream into a smaller, (somewhat) lower quality Vorbis stream, and does so quickly, by just throwing out some data.] After having prototyped several peelers that aim to peel to a certain filesize, or to a certain quality, with mixed success, I've now taken a different route: a peeler that aims for the

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?

> 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:

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

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 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

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