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ABX Just Destroyed My Ego, My perception of my bitrate needs was greatly inflated.
audiomars
post Jun 14 2006, 08:47
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@ dpaint4, welcome to the real world laugh.gif laugh.gif

I always thought I could distinguish easily between bitrates and quality but when I really tried to ABX a few tracks, I fell flat (not to mention the severely deflated ego) biggrin.gif

I realised that my ears are not all that "golden" after all and I depend on the (established) golden ears at HA for any quality comparison. Enjoy the music, friend, and if a lower bitrate is alright for you, that's good 'cos you can take more of that wonderful thing with you!!

audiomars
(He who enjoys his music more now that he does not try to find artifacts in every music sample)

This post has been edited by audiomars: Jun 14 2006, 08:49


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legg
post Jun 14 2006, 16:19
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QUOTE (ShowsOn @ Jun 13 2006, 22:41) *
QUOTE (legg @ Jun 14 2006, 13:13) *

That's funny, I use the word watery to refer to warbling.

Woudln't just saying warbling be better? The problem with "watery" is that it means different things to different people, it isn't a description of an artifact.


It would be, if I the primary language in Mexico was english. A "watery" equivalent in spanish carries more meaning than warbling. At least to common people.


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dpaint4
post Jun 14 2006, 16:48
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QUOTE (audiomars @ Jun 14 2006, 02:47) *
Enjoy the music, friend, and if a lower bitrate is alright for you, that's good 'cos you can take more of that wonderful thing with you!!


My findings have pushed me towards a kind of unexpected decision, and that's that I plan to keep my lossy library in LAME 3.97b2 at V5, which I previously considered to be 'intollerable'. But now that I know my actual limitations I'm going to enjoy them by having this rather comact library in a very compatible format.

Despite how much AoTuv amazed me (it's now my undisputed favorite), I think I prefer to have one lossless and one lossy library. Keeping three librarys is a bit much. And LAME was pretty darn impressive too.

Of course, it does hurt pretty bad that it has taken me a couple years of exploring new codecs and settings, and building up this huge image of what is acceptable compression and what isn't, and subscribing and unsubscribing to Audiophile magazine (they rarely talked about compression or anything much more than analog and compact disc) to arrive at the one codec and setting that most 'ignorant consumers' use without thinking.

But at the end of the day I guess I'm glad!
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Emanuel
post Jun 14 2006, 17:17
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QUOTE (dpaint4 @ Jun 14 2006, 04:54) *
QUOTE (sld @ Jun 13 2006, 15:36) *

dpaint4, what's your laptop's soundcard? A better external soundcard may help, unless the existing one is from m-audio or something.


I'm absolutely sure that my laptop has some kind of integrated sound thing. It's seriously not high end. But I don't think I can hide behind that excuse. Certainly my laptop sounds as good as my iPod or my X5. And it's better than my last computer which had awful ambient noise. This one is at least silent when it's supposed to be.

// Take my note only as a personal reflection, not a statement //
I have a hard time abx:ing 128 kbps in both recent mp3 encodings and in vorbis using my laptop integrated Soundmax chip. It gets a lot easier when I use the M-Audio Firewire 410 (both with AKG 240 M headphones). The best description I can give, is that the Soundmax sound is kind of "blurry". So, do this test of yours again using a better soundcard!


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detokaal
post Jun 14 2006, 22:29
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QUOTE (Gambit @ Jun 13 2006, 15:59) *
I think the reason is in large due to the common misconception that audio compression heavily alters the sound. Less dynamics, weaker bass and all those other descriptions "audiophiles" like to throw around, and that in fact are nothing more than just placebo. But in reality, the artifacts are much more subtle, and often require actual training for an inexperienced user to be able to hear them. So when somebody tries an ABX test for the first time, without previous training the results are most of the time surprising.

But I would say that sometimes it's not really a good idea to train for compression artifacts. I guess in this case you could really say that ignorance is a bliss. Enjoy your music and forget about the golden ears.



Ignorance IS bliss. It used to be that only changes in tonality bothered me - as a classical and jazz musician very familiar with timbres of all types of instruments and ensembles, audio compression easily stuck out in ideal (quiet) listening environments. Then I did some training on pre-echo, stereo, etc. on web sites mentioned here on AH and I regret it because those sometimes now bother me too. My advice? Just enjoy your compressed music as it is and don't worry about ABXing anything.
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molnart
post Jun 15 2006, 00:28
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this topic forced me to do my first abx test:
LAME -V8 was easy but -V7 was already indistinguishable
I have to admit that my equipment is rather poor (integrated SoundMAX chip, cheap Phillips headphones), so i'm still sticking with -V2 for future use wink.gif

I remember when i did my first encodings with Lame 3.92 at 128kbps CBR i was quite satisfied with the results and never worried about the quality, but mp3s from different sources (downloads, friends) even at high 320 kpbs were (and still are) suspicious, not to mention the possible ripping errors.
However the most annoying artifact was always the missing gap-less encoding

maybe i'll get bladeenc and try to abx it at 128kbps, just to compare a modern and old codec


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AstralStorm
post Jun 15 2006, 04:45
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QUOTE (Emanuel @ Jun 14 2006, 18:17) *
// Take my note only as a personal reflection, not a statement //
I have a hard time abx:ing 128 kbps in both recent mp3 encodings and in vorbis using my laptop integrated Soundmax chip. It gets a lot easier when I use the M-Audio Firewire 410 (both with AKG 240 M headphones). The best description I can give, is that the Soundmax sound is kind of "blurry". So, do this test of yours again using a better soundcard!


Indeed, some AC97 chips have cheap resamplers built in.
You should test your AC97, Windows resampler (in case it's done in the software) and M-Audio in ABC/HR+ABX.

Some far from perfect resamplers have too low lowpasses.
E.g. Sensaura resampler removes high frequencies. (ABXed vs PPHS foobar2000 resampler on Ultra)


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pepoluan
post Jun 16 2006, 11:47
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QUOTE (Lyx @ Jun 14 2006, 10:40) *
QUOTE (Ruby @ Jun 14 2006, 01:48) *
Though, must admit that massive hangovers and Britney Spears don't exactly help with ABX biggrin.gif
Thats normal. It cannot sound much worse, even if you encode it lossy.
I have the nagging suspicion that in your case, lossy encoding improves it considerably... laugh.gif


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user
post Jun 16 2006, 13:54
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erm,
the truth is different for different listening envirements and people.

Though I am meanwhile in my 30s even going 40 biggrin.gif biggrin.gif and luckily don't need help yet to walk over a street biggrin.gif biggrin.gif,
lossy formats ABXing at 128k vbr target bitrate are still no problem for me,
"even" not with headphones, but with 3 way speakers.

1 fact is, "even" mp3-Lame made huge improvements with time, especially at 128k vbr -V5 --vbr-new.
The "artifact" though, which let find me lossy material easily, in normal music, which had been recorded/mixed/mastered in god quality: highs, attacks.
So, in past, abxing was easier to me, which might be due to my younger age at that time, and probably by the progress of lossy formats. But it's still easily possible in normal music, no "problem samples" !, at 130k vbr.
I add, that I visited loud concerts very rarely, and after my 1st Jethro Tull, I bought ear plugs to protect my ears in concerts.
I listen sometimes to headphones at home, but mostly via good broad HiFi-speaker setup, source since pc days lossless or hq-mpc 220k vbr and higher, digitally fed to amp.

at topic starter:
your laptops 44.1->48k sampling will already destroy a lot, or even more, the cheapo/crappy analogue outputs of your laptop. Your laptop might improve, if you use digital out, maybe let foobar2000 even resample to 48k on your laptop.

I carry out my fine abxings on stereo hifi speakers, cd-player, with burnt cd-r contaning the tracks in different formats/codecs and the lossless/original track to have an A comparison anchor.
Switching inside 1 track helps, where the same track in another format runs with few seconds difference, so that a passage is repeated.
so, you can carry out some A/B and X/Y listening and ABX if needed.

If I listen to headphones, I used always lower volumes than my friends. see my need to wear ear protections at concerts.
btw., my brother and his friend, who visited some concerts eg. guns n roses, have both tinnitus sometimes...
and yes, my brother thinks, it's just my imagination to need more than 128k bitrates to be transparent.
Though, at certain evenings he was able to listen same differences in music at certain effects with my setup.
He has personally a way smaller/simpler hifi setup.
But he brought me into compressed/pc audio, though at his 1st party, where he played only music as mp3 (most 128k cbr stuff, at very early times in the 90s...), I was hurt by that analogue music output..., without abxing... (probably the cheap soundcard together with very crappy mp3s at party volumes via broad stereo setup, excellent amp and good speakers)
So, lossy audio can be satisfying, transparent even for guys with fine ears and good hifi, if bitrate is roughly doubled to contain nearly full spectrum and make no comprises regarding stereo content, ie. >220k vbr, (imo mpc preferred, as it was especially tuned for this, and this will not change, even as it is some months ago smile.gif )
To put this into relativismn:
I like to listen to 128k vbr lame-mp3 nowadays, in non-hifi environments,
ie. car stereo, or running outdoors with Koss KSC 75 headphones (owning for hifi: AKG K 500)

and: abxing results have nothing to do with ego !
ego = style of music u listen too...


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vinnie97
post Jun 16 2006, 19:13
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Ego certainly is related....because he perceived his hearing was better than it actually was, therefore deflating his ego. wink.gif

QUOTE
(imo mpc preferred, as it was especially tuned for this, and this will not change, even as it is some months ago smile.gif )

Yes very good choice for this bitrate but there are no improvements being made either. Have you tried ABX'ing the latest versions of Vorbis, MP3 and MPC at that bitrate range? I'd love to see your results. (I have problems ABX'ing 128 kbps, so I wouldn't be a good candidate to conduct such a test).
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ryran
post Jun 20 2006, 01:35
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Okay.. uhmm.....

I just did a little abx test.
First I tried to ABX between lame -v2 & v3.
I had kinda figured that would be hopeless.

So next I added in a lame v6 file to the mix. Rrrroight. No cookies.

Next step. I threw a FRIGGEN v8 encode of the same track into the mix.
...
I cannot tell any difference between all four. Wow.

Finally I dropped the v2 and added a v9.
Interesting. I can pick out the v9 everytime, but would just be guessing to pick between the v3, v6, and v8. Man.

SOOOOOOOO, conclusions: obviously, this was just ONE track and so.. doesn't necessarily mean.. well, anything. But somehow I don't think it would be premature to join the I've-been-encoding-my-music-at-way-higher-bitrate-than-necessary-for-my-ears club.

This post has been edited by ryran: Jun 20 2006, 01:38
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pepoluan
post Jun 20 2006, 03:06
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Hear, hear.

/me joined the club too. Although this time I use Vorbis.
-q 2 & -q 3 = Hopeless
-q 1 & -q 2 = I thought I heard something. But still no go.
-q 0 & -q 1 = FINALLY I heard something. If I pay extra attention, that is (1st 5 trials easy, but the next trials up to #16 exponentially more difficult)
-q -1 & -q 0 = Rather easy

BUT

Played the -q -1 file in its entirety... very acceptable. (Not to mention still (sliiiightly) smaller than AAC 48 kbps).

Whoa.

I've a feeling that 48 kbps test by Sebastian is the 'just right' bitrate... and many testers will be blown out of the water when rating the quality.

For me personally, I think I'll lower the -q for my mobile devices.


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Khushrenada
post Jun 20 2006, 03:08
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why not lossy vs original/lossless...?
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uart
post Jun 20 2006, 17:41
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A while ago I tried ABX-ing my mp3's (-V4) with the lossless originals and I seemed to be able to tell them apart about 75% of the time. The funny thing though was that I couldn't really say what I noticed that was different, each time it seemed very like a guess but somehow I got a lot more right than wrong. I still think Lame -V4 sounds good enough to me. I also only want one lossy copy and Lame @ -V4 is a filesize/quality/compatability compromise that works ok for me.
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french dok
post Jun 20 2006, 18:08
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QUOTE (molnart @ Jun 15 2006, 01:28) *
this topic forced me to do my first abx test:
LAME -V8 was easy but -V7 was already indistinguishable

My ears also quite suck. I'm using -V6 with lame and -q0.35 with NeroAAC. But -V7 is for me easy to ABX : indeed V7 is at 32kHz whereas V6 is still at 44kHz, so the difference in high frequencies is quite important.
And I think i'll try to compress a bit more with NeroAAC, because 0.35 is very transparent to me.
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LaserSokrates
post Jun 20 2006, 18:17
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QUOTE
And I think i'll try to compress a bit more with NeroAAC, because 0.35 is very transparent to me.

Transparancy is binary.
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Silversight
post Jun 20 2006, 18:25
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So, here are my Vorbis lossy-lossless ABX results. I used the first seconds of "Shallow" by Poets of the Fall, an acoustic guitar solo, and aoTuV 4.51.

Q 1: Quite easy, a "warbling" sound and a lowpass (at least I think): 12/12

Q 2: The warbling slowly fades away, lowpass still present: 12/12

Q 3: Warbling nearly non-existent, lowpass seems higher: 13/14

Q 4: No more warbling, lowpass nearly non-existent. Ears getting tired, I stop testing here: 8/8, but then 10/13.


I think that, depending on the source material, for me Q 4 or Q 5 are indistinguishable from the original. Well, I did expect that. smile.gif


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Pio2001
post Jun 20 2006, 22:27
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Well, you made me ABX the usual codecs again !

I chose a track that should be hard to encode : Depeche Mode - Some Great Reward - Something to Do (old mastering).

I begain with Lame 3.97b2. I checked the first encoding for clipping and offset, it's all right (I decoded with Foobar2000).

-V5 --vbr-new : 131 kbps, ABX 16/16. Easy (noise)
-V4 --vbr-new : 151 kbps, ABX 16/16. Easy (noise)
-V3 --vbr-new : 163 kbps, ABX 16/16
-V2 --vbr-new : 192 kbps, ABX 16/16. More difficult. The noise is partially masked and stands out as a slight resonance.

Since I thought that I heard a stereo problem (low frequencies seem out of phase), I tried
-V2 --vbr-new -ms : 221 kbps. ABX 16/16

Then I abxed -V2 --vbr-new versus -V2 --vbr-new -ms (the first one gives 17.7 % LR and 82.3 % MS) : 15/16

However, this is not the stereo that made the difference, it's the fact that the V2 -ms encoding sounds as bad as the -V3 one

Next, I will have to try Musepack and Vorbis


EDIT : oops ! Sorry, I scored only 15/16 ABXing the -ms option, not 16/16 !

This post has been edited by Pio2001: Jun 21 2006, 10:01
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ryran
post Jun 20 2006, 22:38
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Good lord, Pio......
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Pio2001
post Jun 20 2006, 23:19
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No, no, no, it's just a killer track ! Or I think so...

Case in point, in the multiformat 128 kbps test of Sebastian Mares, for Lame MP3, I could only ABX 3 of the 8 samples that I tried !
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kennedyb4
post Jun 20 2006, 23:28
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I totally agree in the ignorance is bliss thing. It took me a long time to abx even a few of the -abr 128 files I encoded.

But I did learn to pick some out so I now resort to overkill to help prevent some future revelation forcing me to re-encode my cds.

Thats why I use -V1 --vbr new --lowpass 20. Hard to ever say that these are bad files. smile.gif
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saratoga
post Jun 21 2006, 08:06
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QUOTE (Silversight @ Jun 20 2006, 10:25) *
Q 4: No more warbling, lowpass nearly non-existent. Ears getting tired, I stop testing here: 8/8, but then 10/13.


Just a heads up, but when you ABX, you need to decide how many trials you'll do in advance. If you stop when you're ahead, you defeat the purpose of the test.
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Ihmemies
post Jun 21 2006, 19:12
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I abx'd one 320kbps lossy MP3 (more chances of mistake wink.gif with Sennheiser HD650/Meier Audio Corda Ha-1Mk2/Audigy 2. I wanted to hear if there really are differences between Audigy's and software resampling. I converted the same mp3 to two different wav's, one with 48KHz Secret Rabbit Code converter (best sinc interpolation), the other without any dsp's. I used normal Directsound output plugin with 16bit output, without any dsp's.

I got 7/7. I guess that's enough, altough some page suggested 16 is considered as a good compromise... well, it wasn't a scientific test anyways, and if I really just guessed them all right accidentally, I might as well go to fill some lotto coupons this week...
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Pio2001
post Jun 21 2006, 19:48
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sinc interpolation blink.gif ? Given the awful results that it gives for picture, I don't think that this is the best way to resample ! It causes a tremendous amount of ringing at the cutoff frequency. 22050 Hz is usually inaudible, but in case of non-linear behaviour from an element of the hifi system, it is safer to use a more classic filter.
The standard filters for CD, for example, go from 0 dB at 20 kHz to -140 dB at 22049 Hz (not sure about the -140, but it is something well below -96 dB). Sinc go from 0 dB at 22049 Hz to minus infinite at 22050 Hz.
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John Stimson
post Jun 21 2006, 19:55
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QUOTE (Mike Giacomelli @ Jun 21 2006, 00:06) *
Just a heads up, but when you ABX, you need to decide how many trials you'll do in advance. If you stop when you're ahead, you defeat the purpose of the test.
That really depends on the purpose of your test. If the purpose is to create statistically based scientific evidence, then you are correct. You cannot prove anything by using the "continue until you succeed" strategy. That is similar to doubling your bet at the casino until you win -- while it works in theory (but not in practice -- even Bill Gates isn't rich enough to keep doubling indefinitely) that fact does not prove that the odds on an individual bet are in your favor.

However, an ABX tool can also be used to gauge for yourself whether you can hear the difference between two tracks. In that case, you are trusting your own judgement about what you can hear, and the cumulative results are simply one more piece of information you can take into account. In that case, you are not performing a true ABX experiment. You are simply using an ABX tool to assist yourself in comparing tracks.

In fact, I was doing just that last night. I wanted to determine what quality level to use with the Nero aac encoder. What I found was somewhat surprising, and right in line with what the original poster discovered. With four tracks picked out of my collection ("The World Turned Upside Down" by Karan Casey, "The Sunbed Song" by Level 42, and a Prelude & Fugue from the Well-Tempered Clavier performed by Ton Koopman on harpsichord) I was unable to distinguish from the original at very low quality levels.

At LC q=0.1 the difference was obvious. The high frequencies were muted in the compressed track.

At LC q=0.2 I just couldn't tell when listening to the whole track and flipping from one to the other.

Then I paid a visit to www.pcabx.com and downloaded some of Arny Kruger's test samples which are designed to make life difficult for perceptual encoders. Castanets didn't do it. I still couldn't tell LC q=0.2 apart from the original. But then I tried the nc-2 "harmonic test tone" sample that's designed specifically to break encoders, and could tell the difference at LC q=0.2 easily. LC q=0.3 was harder, but I could still tell the difference. Next I'll have to try LC q=0.4. The interesting thing I noticed was that the average bitrate for the nc-2 compressed sample was about half the average bitrate for the music tracks. So apparently, the encoder's method of assessing the required bitrate doesn't think that sample needs a high bitrate.

I am not sure, but I think that the Level 42 track that I used was the same one that provoked LAME's CBR 128kbps encoder into very easily heard pre-echo when I tried it a year ago. So I am very impressed with the Nero compression, since it isn't nearly as easy to distinguish between Nero at around 85 kbps average and the original.

The question is, If music doesn't display any obvious artifacts, then do I really need to increase the quality parameter until the specially engineered test tone is indistinguishable from the original? I know, that's a personal decision, but one that a lot of us have to make. It sure would be nice if I could use a higher level of compression and fit my entire collection onto my portable player.

I've got a lot of listening to do -- maybe I can find music tracks that are as critical as Arny's test tone. Or maybe I can't. Perhaps the easiest & laziest plan would be to encode everything at LC q=0.2 and wait to see if I hear something that's unacceptable.
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