similar to: applaud.wav sounds worse in 1.0?

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

I finally got RC3 built on my home machine (had to sneakernet in a tarball of libcurl to get it to build :-( ) and started testing it out on the range of quality levels, side by side with lame using it's range of vbr quailty modes. Overall, it sounds pretty good, but I haven't really tested a good range yet, as I ended up starting on udial.wav... vorbis handles it ok up to quality 3,

SUMMARY I'm impressed. RC3 withstood anything I could throw at it. "-q 3" is really Vorbis' sweet spot, almost perfect, and "-q 0" is eminently usable, with only marginal defects for normal usage. Already sent money, will do that again. * * * I spent at least 40 hours in the last week testing RC3. I selected a few fragments from pop, classic and jazz CDs in my

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

I've updated the "Introduction to compressed audio with Vorbis" tutorial based on the many good corrections and suggestions I got here. Changes were: fixed range of 16bit samples (-32768 to 32767 rather than 0 to 65535) added sentence on why 44.1KHz was chosen for CDs added Sony's name as developer of CD format wrong word ("Which" ->

I've updated the "Introduction to compressed audio with Vorbis" tutorial based on the many good corrections and suggestions I got here. Changes were: fixed range of 16bit samples (-32768 to 32767 rather than 0 to 65535) added sentence on why 44.1KHz was chosen for CDs added Sony's name as developer of CD format wrong word ("Which" ->

>>> 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

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 or decoding. Opus remains unbeaten for me at 133 kbps. That's totally awesome. Cheers On 10 April 2017 at 17:14, Agustín Dall'Alba <agustin at dallalba.com.ar> wrote: > Hello! I found a

I tried darkice 0.9 with icecast 1.3.12 and lame 3.91. I specified a 64k stream and similar options to what works for us with liveice. Streaming worked successfully, and I applaud darkice for being fairly simple to build and configure. But the stream sounded like mud. It was worse than our 16k mono RealAudio stream. I re-examined my config, I rebuilt everything with gcc 2.96, then with

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 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

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

I''m new here and I wanna make some marketing to Redcar Editor<http://redcareditor.com> . I have used this amazing editor for a while and I have to applaud it. It''s an excellent free (as in speech) alternative to TextMate. It''s made on Ruby (run on JRuby), so it has the heart of the Ruby community. It has support to TextMate Themes and Snippets, it''s very

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

On 7 Nov 2017 13:36, Lucas Clemente Vella <lvella at gmail.com> wrote: 2017-11-07 11:10 GMT-02:00 encrupted anonymous <sergeinakamoto at gmail.com<mailto:sergeinakamoto at gmail.com>>: did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable

Opus is specifically designed to survive tandeming but you need to keep the frames aligned and not mess with the gain, which your tools probably do not do. On Thu, Nov 16, 2017 at 10:58 PM, encrupted anonymous < sergeinakamoto at gmail.com> wrote: > using iTunes i've noticed that AAC is > very good at re-encoding own lossy sound. > let's test Opus! > > neroaacenc.exe

Anybody know where I can find ABX samples (original and ogg) that used RC3 or better encoding? The only ones I've been able to find are RC2 (or RC1) PS - on a side note, on pcabx.com's "very easy" training samples, I seem to be completely deaf to the differences between the "Original" and their "test" samples. (Distortion sample only), but I do hear

I'm trying to setup a samba4 with openchange which uses postfix+dovecot+openldap on a ubuntu 12.04.2 system. I can send outgoing emails but incoming emails fail. I looked up the logs and I get: Mar 5 11:03:48 testerA dovecot: lda(sammy): Error: user sammy: Initialization failed: mail_location not set and autodetection failed: Mail storage autodetection failed with home=/var/mail Mar 5

Hi all, I've got a strange problem. I was just encoding a new CD I bought a couple of days ago (Muse - Hullabaloo Soundtrack) when I ran into a strange bug. After ripping track 8, CD 1 with cdparanoia under OpenBSD/alpha, I wanted to encode it with oggenc (latest version from the OpenBSD-current ports tree, (`oggenc -v` says 'OggEnc v1.0 (libvorbis 1.0)')). Here's the output :