similar to: Interactive RC3 quality analysis graphs

Well, this question comes once and again. I have RC3 and Garf's RC2 tuned versions (160 and 350) in a nice graph. Nobody should be surprised that RC3 is better, but in some cases GRC2 shows better EAQUAL/bitrate results. There seems to be room for improvement in RC3 with some tweaking. See it by yourself at http://audio.sinderman.com/ Cheers, AGS.

I was able to ABX 60.wav from the lame samples up to q=5.9 (13/16 tries at that quality level). I failed to ABX it at q=6.0 (8/16). The sample exhibits a pumping artifact; a "fluttering", if you will, of the amplitude. The effect becomes more pronounced as quality setting / bitrate decreases. Any thoughts? Also - if anyone replies to this, could you CC me a copy? I'm not on the

Hello! I found a sample I can ABX successfully when encoded at 133.333 kbps. I was targetting 1 MB/min. https://drive.google.com/drive/folders/0B8KWShoIrA1kQzR1Z0FFRUlfcEU floex.wav is 4:54–5:04 of a lossless copy of 'Forget-me-not' by Floex, downloaded from http://store.floex.cz/album/zorya floex-133.opus was created with `opusenc --bitrate 133.333333 floex.wav floex-133.opus`,

I have run an ogg analisis using EAQUAL from -q 0 till -q 10 evaluating bitrate and quality. The results are shown in a nice graph, and can be seen here: http://audio.sinderman.com/ One thing I have found is a bug, as it is cleary shown by the blue lines. Other discoveries are written in that page, and others are still evolving in my mind :) The goals of this analisis is to help the developers

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > I halved the volume of the sample before encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding

Hello. First of all, thanx for that great codec. I would encode all my music with Ogg Vorbis in the future, if it hadn't been for some samples I encoded for test purposes: Everything was great as long a I used "natural" sounds/music, but certain pieces of synthesized music (specially techno) contained very audible artifacts even at high bitrates. So I just went on using mp3 for

Hello! First of all, 100x thanks to Monty and colleagues: you have done an excellent job! I just didn't believe my ears when I first tested Oggenc 1.0 at q0 to q1 - it sounds AMAZINGLY GOOD !!! But as HDD drives are getting larger and cheaper, most of us move toward higher quality settings ......... I use q8, because: - I was able to ABX some test samples up to q4.99 - at q8 Ogg is still

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

Interesting. Not what I expected. This is Ruby 1.8.6. > irb irb(main):001:0> str = "abx\bc" => "abx\bc" irb(main):002:0> str.length => 5 irb(main):003:0> s = str.sub(/.\b/, '''') => "ab\bc" irb(main):004:0> s.length => 4 irb(main):005:0> s = str.sub(/.\x08/, '''') => "abc"

I want to extra the part of the formula not including the response variable from an lm object. For example if the lm object ABx.lm was created by the call ABx.lm <- lm( y ~ A + B + x, ...) Then ACx.lm is saved as part of a workspace. I wish to extract "~ A + B + x". Later in my code I will fit another linear model of the form z ~ A + B + x for some other response variable z. I

Hi! I have a problem of curve fitting. I use the following data : - vector of predictor data : 0 0.4 0.8 1.2 1.6 - vector of response data : 0.81954 0.64592 0.51247 0.42831 0.35371 I perform parametric fits using custom equations when I use this equation : y = yo + K *(1/(1+exp(-(a+b*ln(x))))) the fitting result is OK but when I use this more general equation : y = yo + K

I have a text processing problem I'm hoping someone can help me solve. This issue it this. I have a character string in which I need to delete a variable number of characters from the string. The string itself contains the number of characters to be deleted. The number of characters to be deleted is proceeded by either a "+" or a "-". A toy example: Suppose I have

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

This is off-topic, mostly, but I figure you guys will have some knowledge in this sound-quality-related area. I'm sitting here looking at the most recent "Musician's Friend" at headphones and thinking about getting a pair. They've got products from AKG, Fostex, Audio Technica, Nady, Sennheiser, and Sony, at price points ranging from $16 to $130 (list prices $20 to

Declan Kelly <flac-dev at groov.ie> wrote: > On Thu, Mar 08, 2012 at 09:50:20AM -0800, giles at thaumas.net wrote: >> I wouldn't worry about it though. It's unfortunate the dbPowerAmp >> developers want to take advantage of the subset of customer who don't >> understand what 'lossless' means. > > I read some of the articles on

I've been looking around for a tool to compare the quality of different audio files, didn't find anything, and so botched together something over the past few days: http://www.btinternet.com/~jfchapman/files/mcp.zip (Win32, 266Kb) It's just a small tool for playing up to 3 files simultaneously, with the ability to switch between the outputs to compare the sound quality (formats

Hi, maybe this is of interest for someone out there. I found that taking the difference between an original sample and its decoded Vorbis counterpart is a cool way to mangle sounds (especially voice samples get a nice weird touch :] ). A side effect is that the result is not only exactly what information the encoder omits, it also contains things that it adds to the sample (artifacts). Leaving

I've been running through a collection of hard to encode samples that the lame developers use, and others, doing ABX testing at various quality levels looking for my personal sweet spot. Last time I did this for RC3 I found that most samples where pretty transparent around the 3.5-4 region (my ears aren't that well trained, and I don't want them to be :-), but this time I noticed

Hi guys, as MP3 and Opus have very similar objectives, I think the original poster's question was a valid one: Why does Opus have more artefacts in the lower frequency ranges than MP3 has? The spontaneous suspect that lower frequency artefacts may be more noticeably than higher frequency artefacts seems plausible, also. Is it a matter of energy (which is higher for higher frequencies)? When