No really. It is useless and a lousy hack. It's just a way for Thomson and FhG to further milk the mp3 buzzword, one more time.
Useless format:
* The lossless part is stored in ID3v2 tags.
* Size of ID3v2 tags is limited to 256MB by specifications; as a result, lossless part of an mp3hd file can't be larger than 256MB.
Addendum: Current tagging software isn't prepared to deal with this kind of situation, so you're going to see various disturbing behaviors such as:
* Very slow tag updates (near-full-file-rewrite with each edit).
* Heavy memory usage of tag editors.
* Retagging stripping correction data.
* Tag editing or even reading failures when approaching the 256MB limit because software will try to put each ID3v2 frame in a single memory block and allocating a single block of such size is likely to fail in 32-bit address space because of fragmentation issues.
I checked the price of an album on iTMS and Amazon before buying it on iTMS. It was five pounds cheaper on iTMS. Note also that there is a huge difference between 128Kb/s AAC encoded with the Dolby Pro encoder (used by iTMS) and the consumer encoder (used by QuickTime). I encode my music at 256Kb/s AAC when I'm using the consumer encoder, and the tracks I've bought from iTMS sound at least as good (you start to get into diminishing returns with AAC around 160-190Kb/s usually).
CD quality is a highly subjective thing. I have encoded a CD as AAC and played it on my iPod and a CD player (with the same amp and speakers) and found it sound better on the iPod (the CD player was from the '80s, and did not handle digital to analogue conversion as well as the iPod. These results are probably not repeatable with a modern CD player).
Sorry, but AFAIK the tracks supplied by iMTS are encoded with the same encoder that is in QuickTime. Can you prove me otherwise? This has been discussed on Hydrogenaudio and, so far, the conclusion is that they're exactly the same.
Oh, and btw, if the source is the same and the encoded file sounds better to you, then there's something wrong with the encoder, cause the objective is to sound as the original as much as possible. Not to "sound better". That's probably placebo effect. Try doing an ABX test.
Check the latest multiformat 128 kbps test. MP3 was tied with AAC (AAC only had a slight advantage, but technically they were tied, check the error margin). Here you can see the end results.
It's completely false that AAC 128 kbps delivers "CD quality" and this test also showed that. No lossy format can do this (yet?). And I'm not talking about problematic samples here. Try to do a simple ABX test (you don't need extra expensive hardware, just some decent sound card and headphones).
It's possible to achive transparency with lossy formats on more than 99.9% of the cases (or whatever), but not with 128 kbps.
These are the encoders used by online stores. iMTS uses the AAC encoder used in this test. No other MP3 encoder can do better at 128 kbps than the setting used in the test with Lame. WMA9 Std was used in this test rather than WMA9 Pro. Why? Because all online stores use this version, and not Pro version, which is better. And the Pro version also lacks hardware support and MS isn't pushing it also because of that (thru WMP9 you can only encode to WMA9 Std.). I don't care, I don't use WMA...
"The unusual quality settings for MPC and Vorbis were chosen after testing several qualities over a wide range of albums and styles, and picking the setting that generated results closer to 128kbps."
Also, as already been said, those numbers are not the average birates of all the samples.
The purpose of vbr is to save bits where the vbr algorithm sees fit.
No, the purpose of VBR is to maintain a constant quality level. And LAME's VBR model is quite well tuned, be sure of that. Try the --alt-preset standard/extreme presets. Please do some ABX tests with some decent headphones (you don't need extra expensive "audiophile" stuff).
About your LAME command line. Up until recently (v3.95 or v3.96, dunno for sure) -q0 could be in fact bad for quality. Once again, check HA and use the search. According to HA, the recommended LAME version is still 3.90.3. You can find it at www.rarewares.org (maintained by rjamorim, who conducted this test). But v3.96 seems quite good too (it has been tested in HA, and the results have been mixed). But please, once again, try the presets.
I'd wonder why songs are encoded from 16 bits / 44100 hz, while our ear can hear a difference with 96 khz/32bits and consumer sound cards can often do 48khz ? (if you can do better than rip from a CD, that is)
I mean, would 160 kbps sound better from 44100 or 48 / 96 khz ?
It's not really proved that we need a sampling rate higher than 44.1 kHz with 16 bits per sample resolution. And don't confuse kbps with kHz. Here's some threads quoted from the Hydrogen Audio FAQ.
There is a lot of corporate FUD about this and sometimes it's hard to distinct it from good tests. Also, we have a lot of bad sounding CDs because they are badly mastered (search for Loudness Race). Some hybrid SACDs even have a badly mastered copy of the album on the Audio CD compatible layer... I wonder why.
Recently I decided to use lame with cbr 192kbps after comparing to the preset vbr settings (including extreme). I use the settings: --cbr -b 192 -h -q0
Using vbr I can hear the noise floor being modulated e.g. by a large amplitude low pass filtered bass sound. I contribute this to vbr changing bitrate. Maybe the psychoacoustic model just doesn't fit my ears:-)
The vbr files average around 200kbps anyway, so they're not smaller than 192kbps cbr.
YUO, sir, are completely clueless about this. *ALL* tests made up to now show that what you wrote is completely false, specially if you are using 192 kbps. Check HydrogenAudio, use the search. Countless tests showed that the CBR approach is completely stupid (unless to comply to some system requisites) if you want to have a constant quality level. That's the way all modern lossy encoders work (be it audio or video). The only exceptions are very low bitrates and, of course, LAME's -alt-preset insane, which uses 320 kbps CBR.
Even most "pirate" release groups have finally adopted VBR for their releases (LAME's --alt-preset standard).
OK, you have a point. But check VorbisHardware for hardware with Ogg Vorbis support. Also, though Lame did well, MP3 is known to have some limitations. But if you have to use MP3, experiment --alt-preset standard in Lame. It was made to offer very good sound quality in bitrates that average below 200 kbps in most cases.
Regarding the results... It's a bit surprising that this third party tuning/tweak of Vorbis did so well. Which is great and I think Xiph should think about incorporating this work on their official encoder as soon as possible, in order to take advantage of its potential. You may be surprised about the relative low performance of AAC. This is partially due to the fact that the chosen AAC encoder was a CBR only encoder (because it was the best AAC encoder at this bitrate on a previous test - Nero encoder is also a good one and offers VBR encoder). With a good implementation of VBR AAC, it should be possible to get a better performance.
While most of the tested codecs/formats showed good performance at 128 kbps, this test alone shows that none can give transparency ( transparency == unability to distinct from the original source for most people and under good conditions) at this bitrate, contrary to what many think. People who think this is important should demand higher quality files from famous online music services (like iTunes Music Store).
People interested in lossy audio encoding should also try Musepack (file extension.mpc). It is considered by many of the hydrogenaudio enthusiasts as the best format at medium/high bitrates, offering transparency with bitrates normally lower (with standard preset ~170 kbps, typical 142... 184 kbps) than what is possible with other formats/codecs. It's now open source (LGPL, iirc). Its biggest disadvantage is the lack of support in portable players (though decoding musepack is faster than decoding the other formats in this test). There are plugins for almost every software player and foobar2000 (which I consider the best one) has native support for it
Unlike other distributed computing projects, Folding@home is run by an academic institution (specifically the Pande Group, at Stanford University's Chemistry Department), which is a nonprofit institution dedicated to science research and education. We will not sell the data or make any money off of it.
Moreover, we will make the data available for others to use. In particular, the results from Folding@home will be made available on several levels. Most importantly, analysis of the simulations will be submitted to scientific journals for publication, and these journal articles will be posted on the web page after publication. Next, after publication of these scientific articles which analyze the data, the raw data of the folding runs will be available for everyone, including other researchers, here on this web site.
Folding@Home isn't run by a corporation (unlike United Devices), doesn't use patented algorithms (unlike distributedfolding) and has already published results (unlike everyone else). And it has a set of realistic goals without Aliens, AIDS and SARS hype.
Ogg Vorbis is now the third most used lossy format (in audio only, not counting movie rips and etc..), following mp3 and wma.
MP3 is losing users to other formats, such as Ogg Vorbis, Musepack and AAC. Everyone watching P2P (:O) networks can note this. There're already release groups that only release in one these formats (Ogg Vorbis or Musepack).
Disadvantages of MP3:
- MP3 can't be gapless. - No standard tagging system (id3 is not in the specs). - Ogg Vorbis and Musepack have way better VBR so less bits are wasted where they are not needed and VBR limited to 320kbps. - No sample-accurate seeking.
Slashdot Mirror
No really. It is useless and a lousy hack. It's just a way for Thomson and FhG to further milk the mp3 buzzword, one more time.
Useless format:
* The lossless part is stored in ID3v2 tags.
* Size of ID3v2 tags is limited to 256MB by specifications; as a result, lossless part of an mp3hd file can't be larger than 256MB.
Addendum:
Current tagging software isn't prepared to deal with this kind of situation, so you're going to see various disturbing behaviors such as:
* Very slow tag updates (near-full-file-rewrite with each edit).
* Heavy memory usage of tag editors.
* Retagging stripping correction data.
* Tag editing or even reading failures when approaching the 256MB limit because software will try to put each ID3v2 frame in a single memory block and allocating a single block of such size is likely to fail in 32-bit address space because of fragmentation issues.
From: http://www.hydrogenaudio.org/forums/index.php?showtopic=70548
Sorry, but AFAIK the tracks supplied by iMTS are encoded with the same encoder that is in QuickTime. Can you prove me otherwise? This has been discussed on Hydrogenaudio and, so far, the conclusion is that they're exactly the same.
Oh, and btw, if the source is the same and the encoded file sounds better to you, then there's something wrong with the encoder, cause the objective is to sound as the original as much as possible. Not to "sound better". That's probably placebo effect. Try doing an ABX test.
Cya
Erm. That's not true.
Check the latest multiformat 128 kbps test. MP3 was tied with AAC (AAC only had a slight advantage, but technically they were tied, check the error margin). Here you can see the end results.
It's completely false that AAC 128 kbps delivers "CD quality" and this test also showed that. No lossy format can do this (yet?). And I'm not talking about problematic samples here. Try to do a simple ABX test (you don't need extra expensive hardware, just some decent sound card and headphones).
It's possible to achive transparency with lossy formats on more than 99.9% of the cases (or whatever), but not with 128 kbps.
Cya
These are the encoders used by online stores. iMTS uses the AAC encoder used in this test. No other MP3 encoder can do better at 128 kbps than the setting used in the test with Lame. WMA9 Std was used in this test rather than WMA9 Pro. Why? Because all online stores use this version, and not Pro version, which is better. And the Pro version also lacks hardware support and MS isn't pushing it also because of that (thru WMP9 you can only encode to WMA9 Std.). I don't care, I don't use WMA...
cya
If you had read the results page you'd see:
"The unusual quality settings for MPC and Vorbis were chosen after testing several qualities over a wide range of albums and styles, and picking the setting that generated results closer to 128kbps."
Also, as already been said, those numbers are not the average birates of all the samples.
That's why I wrote "The only exceptions are very low bitrates and, of course, LAME's -alt-preset insane, which uses 320 kbps CBR."
:-)
;-)
Happy now?
DOH, I shouldn't reply to trolls.
No, the purpose of VBR is to maintain a constant quality level. And LAME's VBR model is quite well tuned, be sure of that. Try the --alt-preset standard/extreme presets. Please do some ABX tests with some decent headphones (you don't need extra expensive "audiophile" stuff).
About your LAME command line. Up until recently (v3.95 or v3.96, dunno for sure) -q0 could be in fact bad for quality. Once again, check HA and use the search. According to HA, the recommended LAME version is still 3.90.3. You can find it at www.rarewares.org (maintained by rjamorim, who conducted this test). But v3.96 seems quite good too (it has been tested in HA, and the results have been mixed). But please, once again, try the presets.
cya
It's not really proved that we need a sampling rate higher than 44.1 kHz with 16 bits per sample resolution. And don't confuse kbps with kHz. Here's some threads quoted from the Hydrogen Audio FAQ.
There is a lot of corporate FUD about this and sometimes it's hard to distinct it from good tests. Also, we have a lot of bad sounding CDs because they are badly mastered (search for Loudness Race). Some hybrid SACDs even have a badly mastered copy of the album on the Audio CD compatible layer... I wonder why.
Even most "pirate" release groups have finally adopted VBR for their releases (LAME's --alt-preset standard).
I'll save you some effort:
Recommended Lame settings
LAME 192 CBR vs LAME aps
why 192kbps?
CBR vs VBR
Please read'em all. Of course, you can use whatever you like. Just don't spread BS around.
OK, you have a point. But check VorbisHardware for hardware with Ogg Vorbis support. Also, though Lame did well, MP3 is known to have some limitations. But if you have to use MP3, experiment --alt-preset standard in Lame. It was made to offer very good sound quality in bitrates that average below 200 kbps in most cases.
.mpc). It is considered by many of the hydrogenaudio enthusiasts as the best format at medium/high bitrates, offering transparency with bitrates normally lower (with standard preset ~170 kbps, typical 142 ... 184 kbps) than what is possible with other formats/codecs. It's now open source (LGPL, iirc). Its biggest disadvantage is the lack of support in portable players (though decoding musepack is faster than decoding the other formats in this test). There are plugins for almost every software player and foobar2000 (which I consider the best one) has native support for it
Regarding the results... It's a bit surprising that this third party tuning/tweak of Vorbis did so well. Which is great and I think Xiph should think about incorporating this work on their official encoder as soon as possible, in order to take advantage of its potential. You may be surprised about the relative low performance of AAC. This is partially due to the fact that the chosen AAC encoder was a CBR only encoder (because it was the best AAC encoder at this bitrate on a previous test - Nero encoder is also a good one and offers VBR encoder). With a good implementation of VBR AAC, it should be possible to get a better performance.
While most of the tested codecs/formats showed good performance at 128 kbps, this test alone shows that none can give transparency ( transparency == unability to distinct from the original source for most people and under good conditions) at this bitrate, contrary to what many think. People who think this is important should demand higher quality files from famous online music services (like iTunes Music Store).
People interested in lossy audio encoding should also try Musepack (file extension
quote from folding@home FAQ:
Folding@Home isn't run by a corporation (unlike United Devices), doesn't use patented algorithms (unlike distributedfolding) and has already published results (unlike everyone else). And it has a set of realistic goals without Aliens, AIDS and SARS hype.
foobar2000
correction to one of the disadvantages:
- Ogg Vorbis and Musepack VBR is NOT limited to 320 kbps... that's why they're better.
cya
Ogg Vorbis is now the third most used lossy format (in audio only, not counting movie rips and etc..), following mp3 and wma.
MP3 is losing users to other formats, such as Ogg Vorbis, Musepack and AAC. Everyone watching P2P (:O) networks can note this. There're already release groups that only release in one these formats (Ogg Vorbis or Musepack).
Disadvantages of MP3:
- MP3 can't be gapless.
- No standard tagging system (id3 is not in the specs).
- Ogg Vorbis and Musepack have way better VBR so less bits are wasted where they are not needed and VBR limited to 320kbps.
- No sample-accurate seeking.
cya
The current implementation (12 MHz) is limited to 64 kbps. For more, a higher frequency is needed.
. ht ml
http://www.xiph.org/archives/vorbis/200307/0242
cya
I mean... SCO claimed code theft and they will have to prove it... DUH!