similar to: AW: AW: AW: AW: Why the commotion about file extensions?

> > That might be true. However, the main problem I see with this is > > that by using only the codecs as the extension you make it > > impossible to filter for audio/video without knowing all the codecs. > > You cannot easily do file searches without specifying all those > > extensions either. > > > Nor can filter audio/video without specifying all of `wma`,

> All true. But note what the user wants according to these definition: > he wants different icons for audio vs. video. Yes, he doesn't know > the difference between `.mp3` and `.ogg`. But that's precisely why > windows hides the extensions from him by default. If he does know, he > would have disabled this hiding. Exactly. The main reason for different extensions from

Well, I only recently subscribed to this mailing list and do not know all the history of this extension debate (I passed up on reading the hundreds of messages that seem to have been written about this topic). I just wanted to make one quick point: from my experience, the average user does not know about file formats, only about content (my experience stems from working for ashampoo

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

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

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

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

> 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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 15.03.2013 12:09, Erik de Castro Lopo wrote: > Marcus Johnson wrote: >> but no matter what you choose to do FLAC MUST REMAIN LOSSLESS. > > That has always been and will remain goal #1 for FLAC. FLAC stands for Free Lossless Audio Codec, so "lossy FLAC" is like "honest politician", or "fast turtle".

Hi Chris, > I just meant -O3 as an example. I'd expect all -O levels to have the > same behavior. -O3 may run passes which are more "lossy" than -O1 > does though, and I'd expect us to put the most effort into making > passes run at -O1 update debug info. I'm not really sure that you could divide passes into "lossy" and "not so lossy"

using iTunes i've noticed that AAC is very good at re-encoding own lossy sound. let's test Opus! neroaacenc.exe -q 0.75 -if 000.wav -of 001.m4a neroaacdec.exe -if 001.m4a -of aac001.wav wavdiff.exe 000.wav aac001.wav Comparing 000.wav - aac001.wav... Max diff: -17.3867dB RMS diff: -33.0851dB Mean diff: -32.4582dB opusenc.exe --bitrate 512 "000.wav" 001.opus opusdec.exe 001.opus

Hello, I am hoping that perhaps you know the answer to my car audio question, or can at least point me in the right direction. I own a 2013 Audi S4 that has the 3G-Plus Multi Media Interface (MMI) with the Bang & Olufsen Sound System. The MMI is able to playback lossy, compressed audio files (MP3, WMA and AAC), but does not have native support for FLAC, ALAC, or WMA Lossless.

Hi folks! Due to the fact that more and more users increasingly use MP3 < 320kbps as their source for encoding music, and publish it as flac files, I suggest that something is done in the flac encoder to possible avoid this. My idea is kinda easy/stupid, but might work; Implement a function that use a FFT to check if the input has frequencies > 16kHz, and informs the user that the file

Cool, thanks for all the great comments. 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. My first suggestion was to use FFT, because I know that 128kbps mp3 have a low-pass

On Fri, Aug 24, 2012 at 4:07 PM, Villmow, Micah <Micah.Villmow at amd.com> wrote: > > >> -----Original Message----- >> From: Villmow, Micah >> Sent: Friday, August 24, 2012 2:56 PM >> To: 'Eli Friedman' >> Cc: LLVM Developers Mailing List >> Subject: RE: [LLVMdev] RFC: Supporting different sized address space >> arithmetic >>

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.