similar to: Why does FLAC not support bitrates?

Dirk wrote: > grep -i bitrate /usr/include/FLAC/* Lossy encoders like MP3, Ogg Vorbis, AAC etc can throw away as much data as is necessary to meet a bitrate goal. FLAC however, is a **lossless** encoder. It makes the smallest file it can without losing any information. Because it has a goalof being lossless, it cannot even attempt to meet a bitrate target. Erik --

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

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

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?

Dear FLAC team, I have some fundamental doubts regarding FLAC encoder: 1) In FLAC encoder there is no option of specifying the bitrate. There is something called compression-level. I believe that based on the nature of input WAV file and compression-level chosen bitrate is coming into picture. As such there is no user control on the bitrate. Is that correct? 2) In FLAC encoder I am also not

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.

Hello, I've got a FreeBSD 6 machine running icecast2 and ices0streaming four different streams of mp3's. Now i have a need to stream the same content as either aac+ or ogg vorbis whichever would be easiest to set up and give the best quality. I want these new streams to handle both broadband and dialup users without sacrificing quality. My problem is i really really do not want to

I started thinking about this after doing a little testing with AAC, MP3 and Ogg Vorbis. I was comparing the different formats at similiar bitrates. After a while I finally realized that they all sound more or less the same to me. >From 160 kbps on, I usually cannot detect any difference between a lossy encoding and the original source. If LAME is used, I have to strain to notice anything at

Hi folks, I'm trying to get a handle on the many lossless audio formats from the perspective of a software developer. I love the FLAC format because it is open source and it performs very well. But there is also the MLP (Meridian Lossless Packing), AAC (Apple Lossless), and probably some Microsoft format(s) that compete in the same feature set. These formats all achieve

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

Hi, I was checking for bit rates support above 500kbps in the libvorbis code, so I tried to encode ogg files with bitrate 576kbps but the encoder OggdropXpd gives maximum bit rate of only 500kbps. Does the Vorbis audio support bit rate 576 kbps? Thanks, Emima -------------- next part -------------- An HTML attachment was scrubbed... URL:

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

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

Suppose, 'fr' is data.frame with columns 'Y', 'A' and 'B'. 'A' has levels 'Aa' 'Ab' and 'Ac', and 'B' has levels 'Ba', 'Bb', 'Bc' and 'Bd'. 'Y' columns are numbers. I tried the following three sets of commands. I understand that A*B is equivalent to A+B+A:B. However, A:B in A+B+A:B is

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

Hi! I am very pleased with the progress that Ogg is making, expecially after I read the latest comparision tests on http://ff123.net/128test/instruct.html that put OGG on top aside with MPC. BUT the APPLAUD.WAV test case ( http://lame.sourceforge.net/download/samples/applaud.wav ) still produces **VERY** easily audible high-frequency artifacts when encoded with OGG RC3 up to q4.9 (!!!). Things

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Hello. I have same problem, and I think that it's because AAC+ uses VBR and when it's changes, is not the same bitrate, because varies... El jue., 9 may. 2019 a las 21:29, Michel van Dop (<mvandop at xs4all.nl>) escribió: > Hi Paul, > > Thank you for the information. I already try a player to discover the > stream type of both streams if the are the same. > But i can

Hello I had an idea about judging the quality of ogg vorbis (or any other lossy codec) I took a wave-file and encoded it to ogg. Then decoded ogg to wav and inverted it's phase. When mixing the original wav with the phase-inverted decoded ogg-file, any identic parts of compressed and uncompressed audio should be eliminated. Of course there's always a "rest" of sound because