similar to: Archival quality for music

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

The last week, after 5 minutes of astonishment for the great compression ratio achieved by ogg (370MB wav of voice -> 14 MB OGG, 28 kbps without any perceivable and/or significant quality loss), I had the idea of use ogg to stream my voice, in order to chat in almost-realtime with a friend (only one, of course: I have a 56k modem). Here the idea: I put a streaming server on my machine (I

ARGH! I am at a complete loss as to which OGG quality settings to use: 8? 10? 3? I'd like to be able to listen to my primarily Rock oriented music on a high-end system (though I don't own one - yet) without any noticeable sound degradation, but I don't want to go total overkill with -q 10. With LAME, I at least used to know 192 kbps with -q0 was a perfect size/quality proportion. I

Hi, I have a question about how to obtain the union of several data frame rows. I'm trying to create a common key for several tests composed of different items. Here is a small scale version of the problem. These are keys for 4 different tests, not all mutually exclusive: id q1 q2 q3 q4 q5 q6 1 A C 2 B D 3 A D B 4 C D B D I would like

Hello, I'm seeking advice here as I'm intend in the not to distant future to start encoding my CD collection, and I'm not sure what to use. Do I use the still current v1.0, or am I better off using a newer build? Also, I've started to notice mention of versions 1.0.1 and 1.1 in these mailing lists. Are these worth holding out for? I'll probably encode at about Q7, and am using

----- Original Message ----- From: "Silvano" <silvano at uel.br> To: <r-help at r-project.org> Sent: Thursday, June 30, 2011 9:07 AM Subject: Tables and merge > Hi, > > I have 21 files which is common variable CODE. > Each file refers to a question. > > I would like to join the 21 files into one, to construct > tables for each question by CODE. >

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

If I'm understanding what's going on in this test correctly, what's happening is: * ARMTargetLowering::LowerCall prefers indirect calls when a function is called at least 3 times in minsize * In thumb 1 (without -fno-omit-frame-pointer) we have effectively only 3 callee-saved registers (r4-r6) * The function has three arguments, so those three plus the register we need to hold the

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

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

('binary' encoding is not supported, stored as-is) Hello, I don't know if you knew that there's a russian site where you can buy your music in Ogg Vorbis. There are many albums where you can decide what Codec (MP3, AAC, WMA, Vorbis, MP+) and Bitrate (or quality for Ogg) should be used to encode and download the files encoded specifically for you. The prices are somewhat low.

From: Jyri Sarha <jsarha at ti.com> I optimized Speex resampler for NEON capable ARM CPUs. The first patch should speed up resampling on any platform that can spare the increased memory usage. It would be nice to have these merged to the master branch. Please let me know if there is anything I can do to help the the merge. The patches have been rebased on top of master branch in

Hello, I have a table (1) of the form q1 q3 q4 q8 q9 A 5 2 0 1 3 B 2 0 2 4 4 I have another table (2): q1 q2 q3 q4 q5 q6 q7 q8 q9 C 10 7 4 2 6 9 3 1 2 I would like to divide the numbers in table (1) by the number of the appropriate column in table (2): q1 q3 q4 q8 q9 A 5/10 2/4 0/2 1/1 3/2 B 2/10 0/4 2/2 4/1 4/2

OS: CentOS 5.0 x86. Hi, I have some old music tapes and a walkman, and want to move the music to mp3 format. So far I am thinking to use Audacity to record the music to wavs, and then do not know, somehow I will compress the wavs to mp3s as vbr 192 Kbps with something. Has anyone done this? What app did you use for recording the music from the tapes and what app did you use to make it mp3?

Dear all, I like the way the Rcmdr package computes reliability. E.g reliability(cov(d[,c("q1", "q2", "q3", "q4", "q5", "q6")], use="complete.obs")) will not only give me the alpha score, but also for each variable, alpha.score if deleted. However - when writing scripts it's very tiresome to load the whole Rcmdr GUI just

I currently have a data frame whose rows correspond to each student and whose columns are different variables for the student, as shown below: Lastname Firstname CATALOG_NBR Email StudentID EMPLID Start 1 alastname afirstname 1213 *@uark.edu 10295236 # 12/2/2008 2 anotherlastname anotherfirstname 1213 **@uark.edu ## 10295236 9/3/2008 Xattempts Q1

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

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

I'm sorry if this question has been asked before; I've looked through the archives and haven't seen anything. The problem I'm seeing is that oggenc's VBR encoding doesn't seem to pay attention to any sort of bitrate limitation, either the -m or bitrate_hard_min settings. It isn't that it temporarily dips below the minimum; the average for the whole (in this case,

In my efforts to convince people to hear how good Vorbis is when compared to MP3, I'm gonna be encoding tracks from my CD collection to both formats, then writing them to audio CD, so people don't have to install extra software to hear the difference. Has anyone got some suggested specs for the options for MP3 encoding when using CDex or Chun-Yu's MP3 writer for winamp (preferably in