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changing file speed with Audacity (audible quality losss ?)
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bomber07
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18. September 2010 @ 20:36 |
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Originally posted by davexnet:
I haven't tried the Audacity stretch myself, I will give it a try.
Some people are more sensitive to 128kbps mp3 than others. Sometimes, to me, they sound
terrible, and other times it's passable. I think it depends on the material
and on whether the file gets any benefit from using joint stereo.
On the face of it, your statement that a 9 times stretch resulted in negligible
degradation sounds surprising - but I would be wrong to judge it without trying it.
Yeah I'm very surprised I can't notice a difference after 9 times, the whole point of me starting this topic was that I was worried there would be an audible difference after just 2-3 times!
And let me tell you I gave it a solid work out!
The original song was 7min9secs, and a very clear clean sounding recording.
I changed the speed percentage +7, -5, +3, -8, +4, -0.3, +0.1, +0.05, -0.03
Back to the time of the original file within .01 of a second.
I'm really struggling to convince myself I can hear a difference.
But yeah the original WAVE vs 128kbps MP3 the difference to me is very clear.
*NOTE - I just tried converting to 128kbps MP3 using dBpoweramp and the difference between it and the original lossless file is nowhere near as clear as when I converted to 128kbps MP3 using Real Player - seems dBpoweramp did a much better job converting to MP3.
I've been meaning to ask, with the Audacity speed change, would you expect audible quality loss to be more obvious when changing a file that was originally very clear and perfect sounding ?, or one that was originally a little imperfect with some noise like hiss etc ?
OK if it is of any interest to you I have uploaded at MegaUpload the exact file I used to experiment with using Audacity speed change, it's the original FLAC from my CD, you can download it here :
http://www.megaupload.com/?d=BTQA0LFW
This message has been edited since posting. Last time this message was edited on 18. September 2010 @ 21:47
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Senior Member
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18. September 2010 @ 23:40 |
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OK Bomber07, I'll give your file a try in Audacity and I'll try it in Sound Forge 8
using their routines. In SF8, there are a bunch of settings, for example, "voice",
"music", 'guitar" - each leaves a slight flavor on the stretch which tales you away
from the original.
SF8 is 5 years old. Perhaps Audacity has found an improvement in the process?
I don't expect to get to his until tomorrow now (because my son is using my box).
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bomber07
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18. September 2010 @ 23:55 |
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Originally posted by davexnet:
OK Bomber07, I'll give your file a try in Audacity and I'll try it in Sound Forge 8
using their routines. In SF8, there are a bunch of settings, for example, "voice",
"music", 'guitar" - each leaves a slight flavor on the stretch which tales you away
from the original.
SF8 is 5 years old. Perhaps Audacity has found an improvement in the process?
I don't expect to get to his until tomorrow now (because my son is using my box).
No worries, I'm just curious as to how my ear compares to a more 'expert' opinion (re : how the quality of the Audacity speed changed version reverted back to original speed compares to the original unedited file).
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Mez
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19. September 2010 @ 01:30 |
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Dave,
All 128s are not created equally. Some use advanced compression before the truncation process. These can sound good if the music lacks acoustic instruments. You also have the young who have the ability to hear the high pitch but have not developed the ability to listen carefully.
I am less surprised that you can't hear a difference after 9 times stretch. Lossless is what, 1300 BR and in a double blind test even great listeners can't tell 190 CBR from lossless. You have a massive buffer to ruin before you can hear the difference.
Originally posted by bomber07:
I have a lot of stuff in the first place, so keeping backup of everything takes up too much room.
Thanks for the more detailed clarification of why you are doing this. Still, you might consider backing up a dozen or so tapes to a DVD using Flac or Ape at the highest compression before you work on them. The compressions are surprisingly efficient. Then when you are satisfied with your twiddling, note the time length then take the back up and process it only once. That will preserve the most quality. Are you using the pitch to determine the proper speed?
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19. September 2010 @ 10:26 |
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Have to agree with davexnet, depends on the material and of course the encoder used..
That said, when making statements like this,
"Yes I can hear an obvious difference between a lossless WAVE and a lossy 128kbps MP3 file of the same source for any song" one wonders how you went about testing..ABX?
Just wondering is all, and as davexnet pointed out, we'd be wrong to judge..
Now over @ the HA forums, statements like that will get you a smack-down w/o backing it up...And they will demand your ABX results..
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19. September 2010 @ 18:50 |
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Bomber07 I'm looking at Audacity. What control are you using?
In "effects", I see change speed, pitch or tempo.
I assume you're changing speed or pitch, since incorrect pitch is the main
problem when cassette is played too fast / slow.
I must admit, my testing with "change speed" got good results, does indeed seem
superior to Sound Forge 8.
I came across this problem once myself a few years ago. However, I fixed it in the
analog realm. I decided the cassette deck itself was out, so I opened it
up and adjusted the little screw on the capstan motor to slow it down a little.
It's been fine ever since.
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bomber07
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19. September 2010 @ 19:55 |
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Originally posted by k00ka:
Have to agree with davexnet, depends on the material and of course the encoder used..
That said, when making statements like this,
"Yes I can hear an obvious difference between a lossless WAVE and a lossy 128kbps MP3 file of the same source for any song" one wonders how you went about testing..ABX?
Just wondering is all, and as davexnet pointed out, we'd be wrong to judge..
Now over @ the HA forums, statements like that will get you a smack-down w/o backing it up...And they will demand your ABX results..
Did you not see I then updated / edited my post to say :
*NOTE - I just tried converting to 128kbps MP3 using dBpoweramp and the difference between it and the original lossless file is nowhere near as clear as when I converted to 128kbps MP3 using Real Player - seems dBpoweramp did a much better job converting to MP3.
eg. Real Player conversion to 128kbps MP3 produced very audible loss of quality, dBpoweramp & Adobe Audition conversion to 128kbps is quite hard to tell from the original un-edited file.
This message has been edited since posting. Last time this message was edited on 19. September 2010 @ 20:03
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bomber07
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19. September 2010 @ 19:59 |
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Originally posted by davexnet:
Bomber07 I'm looking at Audacity. What control are you using?
In "effects", I see change speed, pitch or tempo.
I assume you're changing speed or pitch, since incorrect pitch is the main
problem when cassette is played too fast / slow.
I must admit, my testing with "change speed" got good results, does indeed seem
superior to Sound Forge 8.
I came across this problem once myself a few years ago. However, I fixed it in the
analog realm. I decided the cassette deck itself was out, so I opened it
up and adjusted the little screw on the capstan motor to slow it down a little.
It's been fine ever since.
Hello,
I'm using the "Change Speed" button.
I can't always fix the speed on a cassette deck, as I have quite a few live casettes transfered to digital which were friends master casettes etc., I often only have access to the digital transfer from master cassette.
I enjoy my live music!, but sometimes you can find a song which should correctly play at 3:15 minutes plays at something like 3:00min. Which sounds clearly out of pitch / off speed, creeping towards sounding like chipmunks! haha
I'd be interested to know if you can tell the difference between the original un-edited file, especially after 2-3 speed changes.
This message has been edited since posting. Last time this message was edited on 19. September 2010 @ 20:13
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19. September 2010 @ 21:55 |
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Quote:
I just tried converting to 128kbps MP3 using dBpoweramp and the difference between it and the original lossless file is nowhere near as clear as when I converted to 128kbps MP3 using Real Player
You either converted it or you didn't.. which is it?..
Quote:
Real Player conversion to 128kbps MP3 produced very audible loss of quality, dBpoweramp & Adobe Audition conversion to 128kbps is quite hard to tell from the original un-edited file.
Either you can or cannot tell the difference..Which is it?
Sorry if it seems I'm nitpicking..
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bomber07
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20. September 2010 @ 00:02 |
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Originally posted by k00ka:
Quote:
I just tried converting to 128kbps MP3 using dBpoweramp and the difference between it and the original lossless file is nowhere near as clear as when I converted to 128kbps MP3 using Real Player
You either converted it or you didn't.. which is it?..
Quote:
Real Player conversion to 128kbps MP3 produced very audible loss of quality, dBpoweramp & Adobe Audition conversion to 128kbps is quite hard to tell from the original un-edited file.
Either you can or cannot tell the difference..Which is it?
Sorry if it seems I'm nitpicking..
Sorry when I said I 'tried' converting the lossless to 128kbps MP3 I mean I 'did'.
The 128kbps MP3 created with Real Player was awful sounding, clearly degenerated, absolutely no need for any kind of blind test.
I then created another 128kbps (CBR) MP3 using the dBpoweramp programme(I think Lame 3.98r encoder) which I found much harder to tell the difference between it and the lossless original.
I ended up mixing up the 128kbps (CBR) MP3 and the lossless original, so I didn't know which ones they were. I listened to one part of the song for 5 seconds (repeated a few times) and decided which I felt was the lower quality sounding one. I mixed them up 10 different times and listened to a different part of the song each of the 10 times (about 3-4 listens lasting 5secs to each part) and 9/10 times I correctly chose the MP3 as the file which sounded lower quality (the only time I got it wrong I was feeling distracted by the noise of kids outside yelling and bouncing a ball).
I could tell by listening to the high pitched part of vocals, guitar, or listening to the hi-hat.
So yeah when I sit down and concentrate on it I do feel quite confident I can tell a 128kbps MP3 created with decent programme apart from a lossless version.
***EDIT*** I just did the same test using the original lossless file and file speed changed with Audacity 9 times, it was much more difficult for me to decide which was the higher quality version, it took me 2-3 times longer to decide each time, and I only got 5/10 correct.
So it really seems Audacity speed change is doing a good job at keeping the quality. I am very surprised...
I used the song I posted a link to in one of my previous posts.
I'm not sure why I was asked in the first place ?, why do you want to know if I can tell the difference between lossless and a 128kbps MP3?
This message has been edited since posting. Last time this message was edited on 20. September 2010 @ 01:53
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Mez
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20. September 2010 @ 10:27 |
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Just trying to keep you all honest...
I have a real test you, if you have the album "Don't Shoot Me I'm Only the Piano Player" by Elton John, see if you can tell the difference between a 128 and lossless. As far as I can tell, all the extreme highs are missing on both the CD and vinyl. Any acappella will also completely fall within the range of a 128 CBR, probably within a 30 CBR.
I truly doubt that dbPowerAmp was at fault for the lower performance 128. I am not defending it but pointing out it was probably your selection of an encoder and the settings. As I mentioned earlier, some encoders use advanced compression before they truncate others just truncate. At 128 this is very noticeable. AAC, Realmedia and WMA all use the same basic encoder that they have modified, which uses psychoacoustic compression before they truncate the highs. In case you do not know or forgot, that compression is extremely effective at removing data you will not hear. It is lossy but not hear able lossy used in VBRs. Since you could have chosen any of those encoders using dbPowerAmp and did not, I think the fault was yours.
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bomber07
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20. September 2010 @ 20:09 |
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Originally posted by Mez:
Just trying to keep you all honest...
I have a real test you, if you have the album "Don't Shoot Me I'm Only the Piano Player" by Elton John, see if you can tell the difference between a 128 and lossless. As far as I can tell, all the extreme highs are missing on both the CD and vinyl. Any acappella will also completely fall within the range of a 128 CBR, probably within a 30 CBR.
I truly doubt that dbPowerAmp was at fault for the lower performance 128. I am not defending it but pointing out it was probably your selection of an encoder and the settings. As I mentioned earlier, some encoders use advanced compression before they truncate others just truncate. At 128 this is very noticeable. AAC, Realmedia and WMA all use the same basic encoder that they have modified, which uses psychoacoustic compression before they truncate the highs. In case you do not know or forgot, that compression is extremely effective at removing data you will not hear. It is lossy but not hear able lossy used in VBRs. Since you could have chosen any of those encoders using dbPowerAmp and did not, I think the fault was yours.
Keep honest ?
Why would I lie about what I can and can't hear ?
Davexnet asked me the question "can I hear a difference between the lossless and a 128kbps MP3?", maybe Davexnet should specify exactly which encoder/settings etc. he meant because this seems to have gone to the extreme, and maybe getting a little off track from the reason I started this topic about wheter using "Speed Change" in Audacity causes audible quality loss, unless the 2 somehow relate to each other in a way which hasn't been clearly explained to me !?!?!
I did some tests using the file I posted a MegaUpload link to earlier in the thread.
OK when converting lossless to 128kbps using dBpoweramp I just changed the original ENCODER Lame 3.98r CBR (which you suggested does not do a high grade job), and I changed the dBpoweramp settings and CONVERTED to Windows Media 10 Audio, using Windows Media 10 Audio Professional CODEC, the TARGET was Bit Rate CBR
and a ticked 2 Pass Encoding, SETTINGS were 128kbps 44kHz 2channel 16bit CBR.
I believe they are the settings you are suggesting are superior ?
Using those settings I just did a test by mixing up the 128 with lossless, I correctly chose 10/10 times the 128kbps file to be the lower sounding quality file, it was quite easy when listening to the hi-hat, to my ears it didn't sound any better than the original CODEC Lame 3.98r CBR MP3.
I then changed the CODEC to Windows Media Audio 9.2, Quality VBR, 2 pass, 128kbps, 44kHz, stereo VBR and once again I correctly identified the 128kbps file 10/10 times, however I don't understand why the file says it's 143kbps when I chose 128kbps, is it really 143kbps?
Then a 3rd time using CODEC Windows Media Audio 9.2, Quality VBR (2 pass unticked), VBR quality 75, Stereo VBR, the output file says it's 149kbps, I correctly identified the lower quality file 10/10 times.
To be honest I don't see how "variable bit rate" is supposedly higer quality, as I heard some really awful patches where the sound was quite obviously lower quality without even having to compare to the lossless file, and this was supposedly 149kbps.
Oddly the original MP3 encoder I used (which you suggested I was at fault for using) seems to me like it did the best job, and has been recommended by Hydrogenaudio after extensive tests as being the most reliable MP3 encoder......
http://wiki.hydrogenaudio.org/index.php?title=Lame
This seems to have gone off track as I can't see how it relates to if anyone can hear the difference berween a lossless file and the same file Speed Changed with Audacity 2-3 times.
This message has been edited since posting. Last time this message was edited on 21. September 2010 @ 05:30
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21. September 2010 @ 10:05 |
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bomber07, don't take it so darn personal..AFAIC, you're the one that's now off-topic..
Seems like you've already determined, based on your particular test method(s) that you can hear the difference between a lossy file with different encoders/settings etc..Fine, as was stated on the other forum(same topic) that certainly is plausible..
Who are we to dispute, since none of us share ears, eq, listening environment, yada, yada..However, as you well know, one can claim anything on the net..Which is why when claiming test results of say, 10/10 correct, you will be asked to provide double-blind ABX test results..Not so much in this forum but certainly over at HA, which they have, btw..FYI, this nothing new..Many folks can tell the difference between a lossless file and a lame encoded 128kbps VBR..Which is why if you want a transparent setting use a higher one e.g the recommended settings..But it's a good idea for you to conduct a proper ABX test to determine your sweet-spot..
I highly doubt, our good friend Mez, was/is calling you a liar..Just trying to keep us all honest..
Oh, and btw, I love ' The Cure'..
This message has been edited since posting. Last time this message was edited on 21. September 2010 @ 10:18
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bomber07
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21. September 2010 @ 10:36 |
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Originally posted by k00ka:
bomber07, don't take it so darn personal..AFAIC, you're the one that's now off-topic..
Seems like you've already determined, based on your particular test method(s) that you can hear the difference between a lossy file with different encoders/settings etc..Fine, as was stated on the other forum(same topic) that certainly is plausible..
Who are we to dispute, since none of us share ears, eq, listening environment, yada, yada..However, as you well know, one can claim anything on the net..Which is why when claiming test results of say, 10/10 correct, you will be asked to provide double-blind ABX test results..Not so much in this forum but certainly over at HA, which they have, btw..FYI, this nothing new..Many folks can tell the difference between a lossless file and a lame encoded 128kbps VBR..Which is why if you want a transparent setting use a higher one e.g the recommended settings..But it's a good idea for you to conduct a proper ABX test to determine your sweet-spot..
I highly doubt, our good friend Mez, was/is calling you a liar..Just trying to keep us all honest..
Oh, and btw, I love ' The Cure'..
I know he wasn't really calling anyone a liar, I was simply responding to his comment about keeping everyone honest, by remarking why would I lie !?, I'm not taking it serious at all, I was joking just as much as he was... ;-)
Yeah I know a lot of people can tell the difference at 128kbps, A LOT !, that's why I'm wondering why I've ended up trying all these different settings/codecs etc., I'm not sure how it all relates to the reason I originally started this thread, which was - does the Audacity "Speed Change" function create audible quality loss...
I've given my own results of a blind type test, I did not hear any difference (though I can with a 128kbps MP3), so I guess there's not much anyone else can add, unless people have their own experiences with the programme/function...
I'm just not sure what relevance the whole MP3 discussion has?, I didn't start it, because I have nothing to do with MP3's, I never ever under any circumstane ever convert anything to MP3 !
Cheers
- Though I am now quite interested to know some statistics on how many people can detect quality loss by ear getting up to 192-320kbps etc, and what the best encoders/settings are etc.
This message has been edited since posting. Last time this message was edited on 21. September 2010 @ 10:51
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21. September 2010 @ 11:11 |
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Quote:
I never ever under any circumstane ever convert anything to MP3
B/C for many it is transparent or close to the original but much, much smaller in size..Even with today's large capacity storage device(s), my first guess would be, portability..
FWIW, my personal transparent settings(with most samples) is lame -V2 ~192kbps VBR or AAC/m4a quicktime TrueVBR~256kbps, for my portable(s)..Not saying it's the "best encoders/settings", only what works and is transparent from the original(s) to my ears..
And now we really are going off-topic..
Cheers!
This message has been edited since posting. Last time this message was edited on 21. September 2010 @ 11:14
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Mez
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21. September 2010 @ 11:50 |
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By 'keeping you honest' did not mean I thought you were lying. No I am sure you are truthful and know what you are doing except I do think you are a bit confused and have miss quoted me terribly.
I did have fun with all of you because some 128 br music will sound identical to lossless because it lacks high tones that normally get removed with a 128 BR audio.
Except for being slow, Lame is an excellent encoder. It has the least defects and produces several formats to fulfill different requirements. However, at least to me, if you use Black & White film you really shouldn't complain about the lack of color. I tried to explain to you the difference between Lame CBR and Real Media, AAC and WMA CBR. Had you read and understood the link you posted I wouldn't have to explain this again. By the way all the members who joined in this thread are HA members. I know you posted this same query on HA as well.
I will try 1 more time...
Lame has 4 modes I only understand 3 of those modes. CBR, VRB and ABR. CBR compresses by truncating (removing the high pitched tones) to achieve a specific and uniform bit rate. VBR and ABR use psychoacoustic compression (PC) as does Real Media, AAC and WMA. This removes data you will not hear. It is lossy compression without actual quality loss. Because of how it works, PC is very spotty and can't be relied on to meet any specific compression. For this reason, they all employ truncation to deliver a more consistently compressed product. VBR truncates at a specific frequencies associated with a particular setting. See the table labeled Technical details of the recommended settings in your link. Out of all the compression schemes this is the only method that produces a predictable and uniform quality. At the highest setting the quality exceeds a 320 CRB mp3. (This is what I use.) Real Media, AAC and WMA truncate the highs to deliver a constant bit rate. The weakness of this scheme is the frequencies for compression are all over the board, for instance, if there is a moment of silence, PB will reduce the bits to 0 but the CBR will fill the space with empty values to achieve the specified bit rate. If there is a great deal of complexity, such as a violin solo that produces all sorts of resonance tones the truncation has to truncate much more than usual, cutting out much of the nuances. In a nut shell it cuts out the highs when it is most needed. VBR and ABR will leave the silence at 0 bits. ABR will use that reserve in spots were the PC is poor so ABR will not truncate as much of the highs as Real Media, AAC and WMA in other parts of the music where it is needed most so the quality will be slightly better even though the file size will be the same. Real Media, AAC and WMA produce commercial files to be sold or 'rented' that license agreements forced them to 128 BR. They comply and provide better quality than a 'normal' CBR. For these, the time index is always correct while VBRs and to a lesser degree ABRs do not work correctly. I feel that is a small price to pay for improved quality but that is left up to the user. HA claims Lame is virtually free of defects while HA claims Real Media, AAC and WMA have many know defects.
No encoder is perfect they all have pluses and minuses. It is up to the user to decide what issues are important then educate your self to know which provides the product with the qualities you desire.
I have wasted enough time on this thread.
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Mez
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21. September 2010 @ 12:33 |
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I go back on my word for this brief statement K00ka understands me well. Yes I did mean ALL of you. There really are audios that will not degrade even at 128. I was being playful but reminding all of you not to be so free with sweeping statements.
Boomber, note there is a HUGE difference between being able to hear a tone and pick out that tone within music. As you approach the limit of your hearing the tone needs to be increasingly actually, logarithmically, louder so you can hear it. If you can hear it at 100 db that means you can hear it. The extremely high tones are usually harmonics of stringed instruments which are soft. Just because you can hear a tone at 100 db does not prove that you can hear that same tone in music because it is not at 100 db and is being masked by lower tones. The whole concept of PA is that you hear lower tones over higher tones. EX do you hear a violin play when a cannon goes off? No. The cannon masks the violin. The cannon data is minute compared with the violin.
Lastly, the reason mps do not go above 320 BR is humans can't hear what is being thrown out. 192 CBR and 160 VRB are thought to be transparent to adult humans. That is debatable but then you can always try it out yourself.
Thanks K00ka
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bomber07
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21. September 2010 @ 12:40 |
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I think I've been misunderstood.
Not only because I've been reading about MP3's for days when my original topic/question was about how well Audacity Speed Change function performs when exporting as lossless FLAC/WAVE.
Thanks to anyone who posted anyway......
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21. September 2010 @ 12:51 |
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I don't think you've Quote:
been misunderstood
..We just have differences of opinion and preferences..I at least feel, it's been a good thread, thanks!..
Do drop by/post again, as I'm sure you'll have much to contribute..
And you're welcome Mez!!..
Cheers all. :-)
This message has been edited since posting. Last time this message was edited on 21. September 2010 @ 12:53
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Mez
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21. September 2010 @ 22:38 |
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Originally posted by bomber07:
I think I've been misunderstood.
Not only because I've been reading about MP3's for days when my original topic/question was about how well Audacity Speed Change function performs when exporting as lossless FLAC/WAVE.
Thanks to anyone who posted anyway......
Please don't go away mad. We have at bit more fun on this board the HA. More importantly, I hope you think about everything we discussed. Lossy audio is not to be ignored you learned a bit and may be better for it. What you picked up could enhance your understanding of the original project.
I think you know the answer to your original question yes. It will degrade the audio. Will you ever hear the difference? No Probably even if you twiddled it back and forth 100 times the best listener in the world will not be able to tell the difference. Why? There is an enormous gape between the detectable by electronic devices and detectable by the human hearing. That said, since you are a lossless man meaning to me you hold to trying to keep the detectable errors to a minimum shouldn't matter that you can't hear them. I suggest you keep a copy of your original then when you get the timing right, use the original and switch the timing only once that will preserve your data quality as best you can.
It has been a pleasure. Come again!
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bomber07
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22. September 2010 @ 01:30 |
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Originally posted by Mez:
Originally posted by bomber07:
I think I've been misunderstood.
Not only because I've been reading about MP3's for days when my original topic/question was about how well Audacity Speed Change function performs when exporting as lossless FLAC/WAVE.
Thanks to anyone who posted anyway......
Please don't go away mad. We have at bit more fun on this board the HA. More importantly, I hope you think about everything we discussed. Lossy audio is not to be ignored you learned a bit and may be better for it. What you picked up could enhance your understanding of the original project.
I think you know the answer to your original question yes. It will degrade the audio. Will you ever hear the difference? No Probably even if you twiddled it back and forth 100 times the best listener in the world will not be able to tell the difference. Why? There is an enormous gape between the detectable by electronic devices and detectable by the human hearing. That said, since you are a lossless man meaning to me you hold to trying to keep the detectable errors to a minimum shouldn't matter that you can't hear them. I suggest you keep a copy of your original then when you get the timing right, use the original and switch the timing only once that will preserve your data quality as best you can.
It has been a pleasure. Come again!
No worries, I'm not mad at all, never have been... ;-)
One of reasons I stated that I think I might have been misunderstood is because I was told not to take something so personally, and now I'm told not to go away mad, so I think I've been a little misunderstood, but seriously all is fine, there are no problems and haven't been, when you commented to keep honest, I knew you were joking, I never thought it was an accusation, my comment "why would I lie" wasn't supposed to be defensive which I think was the way you perceived it to be...
Even though I have wondered a little how I ended up absorbed in MP3's / 128kbs files for 2-3 days I still enjoyed reading everything everyone had to say, I wouldn't have done various blind tests of 128kbps files / MP3's using different Encoders/Settings etc. if I wasn't interested... I've learned a few things like the limits of my lossless/lossy radar !, I've been aware of it for years but never really pushed any boundaries with tests etc. It's actually got me very interested to see some statistics on what % of people can detect lossy from lossless at the different bitrates etc, something like that would be very intersting to see...
Again when I asked you if the Codec/Settings I switched to where the ones you were suggesting, I wasn't being defensive as it may have seemed, I was genuinely wanting to check with you that I had used the setting you were suggesting, because if you're telling me a specific codec does the job the best, then naturally that's the one I want to make sure I'm using !
Thanks again for all the info...
There's still a couple of things I asked which I hoped someone could please answer which I'm intersted to know eg. when using Lame VBR and I choose 130kbps, why does the output file say 146kbps ?
Another example with a different Codec Windows Media Audio 9.2 setting was 128kbps VBR but the output said 143kbps.
Is this normal or some kind of error ?, and are the files really boosted to 143kbps even though I chose 128 ?
Thanks again it been enjoying/interesting.
PS - When you say most likely Speed Change will not cause audible quality loss because you're working with lossless formats, it confuses me a little, for example on this very page Davexnet seems to say he tried Speed shift with "Sound Forge 8" with inferior results when compared with "Audacity", which tells me that just because you use lossless formats doesn't mean it's very unlikely (safe) that you won't hear any loss, because it seems like Davexnet noticed a difference with the first 2 programmes he compared, which says to me noticeable quality loss is extremely possible depending on which programme you use (unless the difference Dave is refering to was based on technical readings of some type and not actually based on what he could actually hear by ear ?).
However like I said, using Audacity I haven't so far been able to notice a difference when doing a blind test vs original lossless, I would be very interested to know if anyone with more expert ears than myself ever gets around to checking Audacity out.
Cheers
This message has been edited since posting. Last time this message was edited on 22. September 2010 @ 03:29
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22. September 2010 @ 09:08 |
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bomber07, when choosing a Lame VBR setting e.g -V5 the encoder will use more bits(higher) for complex parts in a track and less(lower bitrate) bits in simple parts..So it's normally for it to display say anything between say 120~150kbps..If you play the Lame VBR track you'll see the bitrate fluctuate up and down..
Back to Audacity, I have it installed on my PC's and lappy, but in all honesty I mostly just use it for recording streams and such, and don't do much if any editing..That said, I have no experience using any other editing SW, so not sure what if any differences there could be, especially when working solely with lossless> <lossless conversion/editing..
Technically there should not be any quality loss when working with LOSSLESS..If davexnet noticed "inferior" results using Soundforge, well who's to say he is wrong?..I'm sure the same goes for other Soundforge users that'll say the opposite..Again,'none of us share ears'..And we mustn't forget the 'Placebo effect'..
This message has been edited since posting. Last time this message was edited on 22. September 2010 @ 09:17
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Mez
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22. September 2010 @ 09:46 |
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Originally posted by bomber07:
1) It's actually got me very interested to see some statistics on what % of people can detect lossy from lossless at the different bitrates etc, something like that would be very intersting to see...
2)Again when I asked you if the Codec/Settings I switched to where the ones you were suggesting, I wasn't being defensive as it may have seemed, I was genuinely wanting to check with you that I had used the setting you were suggesting, because if you're telling me a specific codec does the job the best, then naturally that's the one I want to make sure I'm using !
3) There's still a couple of things I asked which I hoped someone could please answer which I'm intersted to know eg. when using Lame VBR and I choose 130kbps, why does the output file say 146kbps ?
Another example with a different Codec Windows Media Audio 9.2 setting was 128kbps VBR but the output said 143kbps.
Is this normal or some kind of error ?, and are the files really boosted to 143kbps even though I chose 128 ?
Thanks again it been enjoying/interesting.
4) Davexnet noticed a difference with the first 2 programmes he compared, which says to me noticeable quality loss is extremely possible depending on which programme you use (unless the difference Dave is refering to was based on technical readings of some type and not actually based on what he could actually hear by ear ?).
I am glad you didn't take me the wrong way.
At AD discussions often go astray and sometimes we are better for it. Mp3s have their place in the audio world, especially the HiFi ones. As the term suggests you can't hear the difference between them and the lossless original if built properly. Although I am a huge lossy proponent, I would never suggest using an mp3 for archival.
The 128 mp3 was probably a sanity check. We do get crazy persons asking questions or making claims. In most cases, even a person with diminished hearing can pick out even the better fidelity mp3s from HiFi. You passed that test with flying colors.
1) Testing would probably be an absolute can of worms! That is why if you ask that question they will tell you to try the testing your self. That is the best way but is very time consuming. One reason is the music picked for the test will skew the results you bumped into the problem with #3. Another problem is, adult hearing is not uniform. The best you will get is posted in the HA lame wiki. 160 VBR and 192 straight truncation are usually about transparent. Every piece is different every person is different.
2) Lame and Helix have the least defects Lame is virtually clean and Helix is nearly as good but is much faster. There is wide agreement VBR is produces the highest quality. The highest settings will give you the best quality. Slow analysis with the Lame encoder will slow the process even more but will improve the quality.
3) You hit the jackpot with this one! It is one thing to read about something and something else when you stumble over it. VBR is all about quality. When you pick a setting you are setting a quality standard and forget about the bits. The estimate is only that, an estimate. I would guess your sample is more complex and less easily compressed than the average audio.
I have personally experienced a range of something about 128 to over 300 bits for converting lossless to mp3 using Lame VBR V-0. Both of which I believed were mistakes and actually got worked up with the low one because I didn't catch it when I made it. I had to go back to the basement and root through my originals to find the original when I discovered 'the mistake' years later. I redid the disk with the same results and I thought the CD was bad. I got a lossless copy of a vinyl audio capture with the same results. The album was the Elton John album I mentioned before. The master lacked the expected highs. The other surprise was classical guitar solo. After you have been using VBR to compress music you will begin to understand the how VBR works. A good recording of a solo stringed instrument will compress a great deal less than that same instrument in a symphony. I know on the outside that does not make sense. There is so much more data in the symphony. However, in the symphony, there are louder and more easily heard lower tones that mask the complex, soft string tones so the result is greater compression.
4. Dave needs to answer that one. Maybe there is more chance of actual change than I believe there should be. What you are playing with may be less complex than what Dave is using. That is why you ought to implement the safest procedure you can. That way you do not screw up some audio capture after doing 25 tapes without a hear able change. I am going to sound like a broken record, converting lossless to VRB and observing the resulting average bit rate you will be able to guess better what music might be problematic due to its complexity.
Lastly, check this out when you get a chance unless you have already done so. This brief video will put perspective on what is detectable and what is hear able. This is really a must see for any serious listener. This provides an explanation why some persons claim they can hear a difference between HiFi lossy while science would indicate otherwise.
Acustic Myth Workshop
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Mez
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22. September 2010 @ 09:46 |
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Double post
This message has been edited since posting. Last time this message was edited on 22. September 2010 @ 09:52
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22. September 2010 @ 10:03 |
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Originally posted by Mez:
Double post
You mean I finally had my first cup of coffee, before Mez?..
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