So the source link is this.
Apple in talks to improve sound quality (http://www.cnn.com/2011/TECH/web/02/22/24.bit.music/index.html?hpt=Sbin)
Can we emulate this technology at all? Are there codecs available that can handle this? I realize that we are not generally able to handle raw 24-bit sound files, since CDs are only encoded in 16-bit, but if these files appear somehow is there an open-source or free codec that can fit our needs in the future? A 24-bit FLAC or MP3 or Vorbis?
And I guess if the answer is no, has anyone ever tried to create this?
I guess we'll have to wait and see what shape these technologies take. Apple already provides more-than-transparent 16-bit AAC files (256k bps), so if they release more-than-transparent 24-bit lossy files we'll probably be left without a source file to encode ourselves. Yet it'd be good to know if there are any alternatives or if anyone has ever taken a look at this before.
Lossy codecs, like mp3, have no inherent bit depth. They will output 24 bit PCM (if the decoder allows it) quite as easily as 16 bit.
Of course, the last 8 bits will probably contain only noise, since it was a lossy encxoding.
And I guess if the answer is no, has anyone ever tried to create this?
I guess we'll have to wait and see what shape these technologies take. Apple already provides more-than-transparent 16-bit AAC files (256k bps), so if they release more-than-transparent 24-bit lossy files we'll probably be left without a source file to encode ourselves. Yet it'd be good to know if there are any alternatives or if anyone has ever taken a look at this before.
Weird question. 24 bit files aren't uncommon. Heck, some people that do online downloads of their music promote the additional depth as an advantage of going the online route. E.g. I have a whole bunch of 24 bit flac files, some are a decade old I think. If you look on archive.org you should be able to find many 24 bit concert recordings in flac. Of course, anyone doing their own production is working in 24 bit— 16 bit is okay for distribution but you don't really want to use it as an intermediate format.
As mentioned, the lossy codecs can handle increased dynamic range. Vorbis and CELT, for example, can provide >200dB of dynamic range (with a suitable floating point decoder), though I'm not aware of any listening comparison showing this to be a real advantage. If you're using a lossy format as an intermediate then the extended dynamic range can simplify gain staging, though lossy as an intermediate is a bad idea.
Can we emulate this technology at all? Are there codecs available that can handle this?
Sure, just use mp3, aac, flac, vorbis, wma, wav, aiff, wavpack, or basically any other format with 24 bit audio.
A 24-bit FLAC or MP3 or Vorbis?
Yes all of those.
Apple already provides more-than-transparent 16-bit AAC files (256k bps), so if they release more-than-transparent 24-bit lossy files we'll probably be left without a source file to encode ourselves.
I thought Apple already claimed to source some of their AAC files from 24 bit sources?
It seemed like, from what I could tell on the wiki at least, that these regular audio formats were limited to a certain Hz range -- below the 24-bit range, anyway. I was wondering if these codecs were not designed to account for this -- much as a Notepad .txt file isn't designed to handle pictures in it. I'm not an audio engineer, so I thought I'd ask.
It seemed like, from what I could tell on the wiki at least, that these regular audio formats were limited to a certain Hz range -- below the 24-bit range, anyway.
24-bit range corresponds to dB, not Hz. Hz is related to sampling rate/2.
Hz and bits are different things. Hz is a unit of frequency, the number of bits determines the resolution of amplitude. 16 bits provide more than enough resolution for the vast majority of tracks Apple has to offer. Of the tracks that may (I said "may", not "will") benefit, this benefit may only be realized in only an extremely quiet listening environment.
EDIT: Garf was faster.
Many models of Mac computers can play 24-bit sound, and the iTunes program is capable of handling such files.
Handling? Yes. Efficiently? Not so much (http://www.hydrogenaudio.org/forums/index.php?showtopic=69879&hl=ALAC).
...and of course, it will be marketed as "HD" audio.
Oh, cool. Does that mean we get to buy all our music over again? Let me see:
Vinyl, CD, Remastered CD, 24bit HD Download. Genius!
C.
The most frustrating aspect of this sort of gimmickry is that it continues to try and blame digital audio in general - and lossy audio, in particular - for poor sound quality, when it's the artists/engineers/producers who are insisting on brickwall limiting et al. that are f*cking up the sound.
I do love that irony, though - 24-bit audio files to distribute music with literally a few dB of dynamic range...such as the kind the esteemed Dr. Dre produces. Oh yes, you *know* you need 24-bit resolution to hear his music how *he* hears it.
Oh, cool. Does that mean we get to buy all our music over again? Let me see:
Vinyl, CD, Remastered CD, 24bit HD Download. Genius!
C.
This is itune land. Remember to include 128kb/DRM and non-DRM upgrade. Maybe the same song on a video ladder: VHS, DVD, blu-ray.
My only real problem, aside from 99.999999999999% of the comments on CNN's website about this article, is that we don't know if these will be 24-bit lossy or lossless files. I would greatly welcome the addition of 24-bit lossless files for the sake of having a lossless archive of music instead of relying solely on lossy audio. I could really care less about 24-bit lossy audio as my own listening tests and experiences show that I am perfectly happy with the 16-bit lossy audio that I have now.
I could really care less about 24-bit lossy audio as my own listening tests and experiences show that I am perfectly happy with the 16-bit lossy audio that I have now.
MP3?
MP4?
These formats are 32-bit float, not 16-bit or 24-bit.
You know what I mean. I have a ton of lossy files sourced from 16-bit 44.1KHz lossless sources that I am happy with. I do not need or want lossy files encoded at a larger resolution.
Maybe it hasn't much sense for average user but for sophisticated listening 24 bits may have sense. I remember some members (including me) presented results of blind tests on 16 vs 24 bits. However later it was discussion about the validity of 16->24 conversion. But that is!!!! The sound engineers and producers are not machines and they take human decisions too. As example there are plenty amount of bad masterings. I have some DVD-audio-s and maybe it doesn't make much sense to go this high theoretically but practically 24 bits source is preferable as it guarantees that there were less stages where some quality degradation can have place.
This is horrible. I was hoping for lossless instead of these marketing geniuses playing the numbers game.
practically 24 bits source is preferable as it guarantees that there were less stages where some quality degradation can have place.
How does it guarantee that the resulting 24-bit audio wasn't just upconverted? If you're gonna trust that, you might as well trust that a 16-bit result was properly edited/mastered.
Sorry, but I think it's already matter of ethics more than sound producing. And, yes, I see a lot of cases when original DVD-audio was just up-mixed from CD version.
I do not need or want lossy files encoded at a larger resolution.
How would you know if you did or didn't? Why would you care? Assuming that the first 15 significant bits are the same between 16 and 24 bit source files, how do you think their lossy counterparts will differ? Have you tested this?
As I see it: another smart way to push up people's perceived need of bandwidth, storage, RAM and CPU power: bu$ine$$ a$ u$ual...
I do love that irony, though - 24-bit audio files to distribute music with literally a few dB of dynamic range
How many bits are needed for music with only a few dB of dynamic range--one?
How would you know if you did or didn't? Why would you care? Assuming that the first 15 significant bits are the same between 16 and 24 bit source files, how do you think their lossy counterparts will differ? Have you tested this?
Why do you care? I don't mean to turn this around but I don't really understand why this is turning into such a big deal. I am happy with the music (both lossy and lossless sourced from mainly 16-bit 44.1KHz CDs) that I currently have. I guess I fail to see what is wrong with that. Now, I don't mean to be a smart ass here but:
Hz and bits are different things. Hz is a unit of frequency, the number of bits determines the resolution of amplitude. 16 bits provide more than enough resolution for the vast majority of tracks Apple has to offer. Of the tracks that may (I said "may", not "will") benefit, this benefit may only be realized in only an extremely quiet listening environment.
EDIT: Garf was faster.
How do you know if 16 bits is enough for most tracks offered by Apple? Have you listened to all of the music that Apple offers through the iTunes Store? Have you tested their 24-bit solution to determine that the benefit is only realized in an extremely quiet listening environment? Have you done that with the majority of the tracks they are offering at the new resolution?
I am simply saying that I am happy with what I have and I don't want any part in 24-bit lossy files. I would partake in 24-bit lossless files just for having a lossless archive but I have issues with buying entire albums and only having a lossy source.
How do you know if 16 bits is enough for most tracks offered by Apple? Have you listened to all of the music that Apple offers through the iTunes Store? Have you tested their 24-bit solution to determine that the benefit is only realized in an extremely quiet listening environment? Have you done that with the majority of the tracks they are offering at the new resolution?
I think there is still no one on HA who has shown through ABX testing that 24bit bitdepth is distinguishable from 16bit when converted properly. The main use case for 24bit data is processing, not end user consumption.
I think there is still no one on HA who has shown through ABX testing that 24bit bitdepth is distinguishable from 16bit when converted properly.
Then it will be interesting at least for me to do some tests (again). Now with proper conversion (if such thing exists)
The main use case for 24bit data is processing, not end user consumption.
And DVD-A is for .... ?
How do you know if 16 bits is enough for most tracks offered by Apple? Have you listened to all of the music that Apple offers through the iTunes Store? Have you tested their 24-bit solution to determine that the benefit is only realized in an extremely quiet listening environment? Have you done that with the majority of the tracks they are offering at the new resolution?
I think there is still no one on HA who has shown through ABX testing that 24bit bitdepth is distinguishable from 16bit when converted properly. The main use case for 24bit data is processing, not end user consumption.
There have been a few studies, no? One I think in 2007 extensively discussed here, that found while higher sample rate than 44.1 didn't matter, people could distinguish 24-bit audio in near silent passages at very high levels. But this requires a very quiet (and likely unrealistic for most people) listening environment, that's probably what greynol was referring to. One of the authors was even posting here IIRC.
The main use case for 24bit data is processing, not end user consumption.
And DVD-A is for .... ?
Money.
My only real problem, aside from 99.999999999999% of the comments on CNN's website about this article, is that we don't know if these will be 24-bit lossy or lossless files. I would greatly welcome the addition of 24-bit lossless files for the sake of having a lossless archive of music instead of relying solely on lossy audio. I could really care less about 24-bit lossy audio as my own listening tests and experiences show that I am perfectly happy with the 16-bit lossy audio that I have now.
I think that this title from Music Week implies lossless downloads - Apple and others plan 'studio quality' downloads
The main use case for 24bit data is processing, not end user consumption.
And DVD-A is for .... ?
Money.
Yeah, and luckily the DVD-A and SACD experiment didn't turn out to well for the money grubbers, so here's hoping that they see it this time.
And DVD-A is for .... ?
more channels, more room (multiple albums on one disk?), copy protection (maybe breakable, but it's not allowed on CD)
I do love that irony, though - 24-bit audio files to distribute music with literally a few dB of dynamic range
How many bits are needed for music with only a few dB of dynamic range--one?
I believe it's Izotope who makes a plugin that can analyze an audio file and show exactly how many bits of dynamic range it actually has/uses.
Someone (I don't believe it was on HA) ran several top 40 songs through it, and some had a 4-bit dynamic range overall - I don't think any were greater than 8-bit.
I do love that irony, though - 24-bit audio files to distribute music with literally a few dB of dynamic range
How many bits are needed for music with only a few dB of dynamic range--one?
I believe it's Izotope who makes a plugin that can analyze an audio file and show exactly how many bits of dynamic range it actually has/uses.
Someone (I don't believe it was on HA) ran several top 40 songs through it, and some had a 4-bit dynamic range overall - I don't think any were greater than 8-bit.
"Bits needed" should trtack dynamic range pretty well, but they are not the same.
Why do you care?
I care that you're misleading people into thinking such things that lossy audio has an inherent bit-depth (it doesn't!) and that there is something inherently different between a lossy file created from a high-resolution master and one created from CDDA and that there is something wrong with the former. So what do you think is wrong, do you think the file will be bigger or that if forced to be the same size the quality will be less, or what? It sounds like nonsense to me.
http://ca.gizmodo.com/5768446/why-24+bit-a...e-bad-for-users (http://ca.gizmodo.com/5768446/why-24+bit-audio-will-be-bad-for-users)
This response is pretty rational, but the comments section is just ridiculous.
It will be interesting to see if they really bring out 24bit files that suddenly sound better as the old 16bit download.
If that happens i hope on every corner of the internet we will see some examples, showing that the same magic was possible with 16bit already and that this 24bit offer is just a way to fool us dump customers once more.
When they start to release 24bit from now on only on new releases there isn´t much to complain if the quality of the 16bit version wasn´t mangled in a way.
And DVD-A is for .... ?
For being awesome. I wish that format hadn't died like it did. So many people listen to music at their computers these days that can benefit from 5.1
And then it comes with a nice packaging setup: you get the DVD-A and all the music videos on one disk.
I care that you're misleading people into thinking such things that lossy audio has an inherent bit-depth (it doesn't!) and that there is something inherently different between a lossy file created from a high-resolution master and one created from CDDA and that there is something wrong with the former. So what do you think is wrong, do you think the file will be bigger or that if forced to be the same size the quality will be less, or what? It sounds like nonsense to me.
I am not implying or misleading anyone, I am simply saying that, right now, this does not interest me unless they are actually offering lossless downloads (again, for archive purposes). That is all. I don't see a point in upgrading resolution unless they are going to offer lossless files. I am not saying that all lossy encoders are stuck at 16-bit, I am saying that I am currently fine with my lossy files sourced from 16-bit lossless sources and I don't think there is a point in offering higher resolution lossy files. I see nothing wrong with saying that especially if someone else is going to pre-judge the entire library offered by the iTunes Store and suggest that most of it would be fine being encoded at 16-bit.
If it wasn't intentional, then whether you mislead someone or not isn't your call. You made a statement that gave me the impression you thought lossy encoders had an inherent bit-depth. You have since made your correction and I appreciate that.
Show me some compelling evidence that iTunes currently offers tracks that are either mastered so low or have sufficient dynamic range to warrant high-resolution audio and I will amend my comment. Otherwise, I believe others have sufficiently defended my point.
Regarding the other point, I can understand that you don't see the need in having lossy files sourced from 24-bit material. I simply question why you specifically said you didn't want lossy files sourced from 24-bit material. How would you know and why would you care?
And then it comes with a nice packaging setup: you get the DVD-A and all the music videos on one disk.
...and how, pray tell, do you get them off that disc, and onto your hard drive, into your music library, and onto your portable media player?
Oh, I know it "can" be done, but it ain't exactly a one-click operation in iTunes...
I simply question why you specifically said you didn't want lossy files sourced from 24-bit material. How would you know and why would you care?
Well I don't want them if I'll be forced to hear people bragging about how amazing their recently repurchased 24 bit music collection sounds
Would it be immoral to be amused with such stupidity?
Would it be immoral to be amused with such stupidity?
I'm not sure, but thanks to Apple we're going to find out
Would it be immoral to be amused with such stupidity?
I'm not sure, but thanks to Apple we're going to find out
Ha ha.
Blu-ray is taking it from where DVD-A and SACD left off, in a way. There is some classical performances on lossless high-res audio, and rock/pop concerts. But I don't think there are albums being put out, or that there will be. Blu-ray is harder to rip than DVD-A though, and its DRM more of a pain in the ass even if you don't rip.
Personally, I as long as there are CDs, I'll be fine. But I don't listen to newer music/artists whose CDs are expensive. Last time I bought a bunch of CDs I bought some for around $4, and the imported ones were like $13.
Regarding multichannel on portables, I'd be very interested to see something like a Dolby Headphone app for the iPod Touch/iPhone. It could work like the VLC app works, loading its own content into the app.
Why is everyone just picking on Apple ? It's not just Apple other companies are following in the same foot steps.
And DVD-A is for .... ?
...demonstrating that there will be no successor to CD as the defacto physical distribution format for music?
This "story" has made the mainstream press in the UK now...
http://www.metro.co.uk/tech/856425-apple-s...-hd-music-files (http://www.metro.co.uk/tech/856425-apple-set-to-launch-hd-music-files)
...and still no details of what it actually means.
Professional music producers generally capture studio recordings in a 24-bit, high-fidelity audio format. Before the originals, or "masters" in industry parlance, are pressed onto CDs or distributed to digital sellers like Apple's iTunes, they're downgraded to 16-bit files.
From there, the audio can be compressed further in order to minimize the time the music will take to download or to allow it to be streamed on-the-fly over the internet.
Why don't record labels at least give retailers the option of working from higher-grade recordings?
"Why?" Jimmy Iovine, a longtime music executive, asked rhetorically. "I don't know. It's not because they're geniuses."
You know what else they have in the recording studio? Multitracks. Now, if they were going to start selling
those at a premium price for people to play with, then I think we could get interested.
Some reports do state it's lossless...
http://www.dailymail.co.uk/sciencetech/art...o=feeds-newsxml (http://www.dailymail.co.uk/sciencetech/article-1359546/Apple-poised-offer-lossless-sound-quality-iTunes.html?ito=feeds-newsxml)
...but this is the UK media we're talking about here, so don't believe anything you read
If they are going to offer lossless, it would be a stroke of genius to do so only at 24-bits. Their customers will have to buy far larger iPods/Pads/Phones as a result.(Assuming that iTunes customers who choose to buy lossless aren't going to bother to convert it to 16-bit lossless because most of those Apple customers see it as a "it just works out of the box" solution and don't mess around with alternatives).
Cheers,
David.
I do love that irony, though - 24-bit audio files to distribute music with literally a few dB of dynamic range
How many bits are needed for music with only a few dB of dynamic range--one?
I believe it's Izotope who makes a plugin that can analyze an audio file and show exactly how many bits of dynamic range it actually has/uses.
This is the essence of lossyWAV. The command line encoder reports the average number of bits removed, but internally it know this value for each block. You can "see" the noise floor (=bits removed) per block by generating a "correction" file and loading it into any audio editor.
You can find audio signals where lossyWAV wants to keep 17 or 18 bits - but not in the top 40 . Just because lossyWAV wants to keep more bits doesn't necessarily mean they're audible. But IME where it choose to remove them it means they're inaudible (unless you're using one of the intentionally not-quite-transparent low quality presets).
Cheers,
David.
Regarding multichannel on portables, I'd be very interested to see something like a Dolby Headphone app for the iPod Touch/iPhone. It could work like the VLC app works, loading its own content into the app.
Unless you have some fancy dynamic headtracked binaural simulation (highly recommended by the way - it's the only way that binaural works properly IMO), then the 5.1>2 conversion is static and can be done at any time. So there's no need for the iPod to do anything other than play stereo files. Just put the headphone mix into those stereo files.
Cheers,
David.
I was thinking it would be good to preserve the 5.1 content in case you're not listening through headphones, but it would probably be hard to get bit-perfect output to a Dolby enabled receiver to decode. Airplay is supposed to do it, and I have had DTS wav being decoded by my receiver through it, but it also had errors.
I also prefer to listen to stereo content through Dolby Headphone though. What I like is that it doesn't try to make it fake surround, it only simulates two speakers in front of you. But to modify the audio on the iPod Apple would have to allow apps to access the music. Not sure if that's allowed (looking it up right now).
The main use case for 24bit data is processing, not end user consumption.
And DVD-A is for .... ?
Money.
And that my friend is what 24 bit iTunes is for. The good news is that their costs for implmenting it will far less than it was to implement SACD and DVD-A. People lost their careers over SACD and DVD-A.
people like engineers, or like marketers and executives?
people like engineers, or like marketers and executives?
Executives
You know what else they have in the recording studio? Multitracks. Now, if they were going to start selling those at a premium price for people to play with, then I think we could get interested.
Dream Theater had that on the collector's edition of their last album:
http://dreamtheater.net/disco_dreamtheater..._linings_boxset (http://dreamtheater.net/disco_dreamtheater.php?s=black_clouds_silver_linings_boxset)
That box set includes a DVD that contains the stem mixes, so you can isolate each band member.
If you want a 24- or 48-track version of an album, then I think you'll have to be on the production team.
I just started buying albums from iTunes last week and already there are talks of potentially higher quality files . I was afraid this would happen, just not so soon.
I will not buy a single song from them until this is sorted out.
As I see it: another smart way to push up people's perceived need of bandwidth, storage, RAM and CPU power: bu$ine$$ a$ u$ual...
If this were the case, we'd see ISPs offering higher speed connections with higher "bandwidth caps" caps to compensate, which are sorely lacking in many parts of the US.
If this were the case, we'd see ISPs offering higher speed connections with higher "bandwidth caps" caps to compensate, which are sorely lacking in many parts of the US.
Unfortunately, this is their business plan. Most of these ISP offers some form of IPTV (such as through VDSL2) too. They don't want people to go outside of their Network with some online third party option such as Google TV, Netflix, etc without them making money.
Well I don't want them if I'll be forced to hear people bragging about how amazing their recently repurchased 24 bit music collection sounds
You don´t haved to wait for that. Already now you can read several confused people writing on the net how "Blue-Ray" 24bit audio sounds against old 16bit "VHS" alike sound
...and of course, it will be marketed as "HD" audio.
Which will go nicely with my HD sunglasses. No... I'm not kidding, lol. They were a present.
I just went back again to the Slashdot thread on this story now that I have 15 more mod points, and was very pleased to see that nothing scoring 3 or above really needed culling. Sure there were some slight factual errors, but the audiophools were successfully kept in check.
Thanks to all of you who helped tell the truth, either through objectivist posts or informed moderation!
I had already been writing a blog entry on post-CD upsells, so I added the Slashdot reference in at the end and published:
cygweb: The Future of Commercial Recorded Music (http://aharden.wordpress.com/2011/02/23/the-future-of-commercial-recorded-music/)
My main point on the 24-bit format upsell is that much of the industry has been loathe to utilize all 16 bits that CD has. Why should we believe that their use of 24-bit formats will mean better audio quality?
Thanks. What do you mean that the resolution of CD is one of the things holding it back?
Regarding the other point, I can understand that you don't see the need in having lossy files sourced from 24-bit material. I simply question why you specifically said you didn't want lossy files sourced from 24-bit material. How would you know and why would you care?
My lack of enthusiasm for 24-bit sourced lossy files comes from the inevitable flood of advertising (especially since we all know Apple likely isn't the only one going to do this) and being required to buy the tracks all over again (whether at full price or not) in order to replace my current 16-bit sourced lossy files that I have already purchased from the iTunes Store. I wouldn't mind doing this if they jump to 24-bit lossless files as I would be paying more for a lossless archive. I don't want to pay more, again, for new lossy files (this would be the second time the iTunes Store started offering new lossy content). They might sound great, I don't know. I just don't want to have yet another lossy "upgrade" pushed on me require even more money. I purchased a handful of tracks back when the iTunes Store was offering 128kbps DRMed music (about $50 worth over the period of ~3 years) and they wanted $0.30 per track ($15 for music I have already purchased) when upgrading to the iTunes Plus standard (which is now what Apple offers music in). I just don't want to pay an additional sum of money for lossy music that I have already purchased. I know that an Apple rep isn't going to hold a gun to my head to make me upgrade but its the whole concept of buying lossy music again that I don't like.
Thanks. What do you mean that the resolution of CD is one of the things holding it back?
I mean that it can't be extended to higher bit depths or sampling rates -- it's not extensible. I said that from a marketing standpoint. I hope it's clear that I'm arguing that CDs and CD-quality audio are satisfactory for most uses and should remain reference formats.
but its the whole concept of buying lossy music again that I don't like.
"24-bit remastered", anyone?
I just don't want to pay an additional sum of money for lossy music that I have already purchased.
I thought you were talking about new purchases. I would have never asked had I thought you were simply stating the obvious. Thanks for further clarifying.
Oh, cool. Does that mean we get to buy all our music over again? Let me see:
Vinyl, CD, Remastered CD, 24bit HD Download. Genius!
C.
Well, no: rights licensing agencies (SIAE, here in Italy) don't chain their permissions to the phisical support or format, but to the opera itself. This means that you can legally download the same operas you've already purchased, IMHO.
In other words you don't infringe anything if you download an MP3 of an original Vinyl that you own, for example...
Or, if in your country you have the fair use, you can also legally claim that you download for backup purposes.
Oh, cool. Does that mean we get to buy all our music over again? Let me see:
Vinyl, CD, Remastered CD, 24bit HD Download. Genius!
C.
Well, no: rights licensing agencies (SIAE, here in Italy) don't chain their permissions to the phisical support or format, but to the opera itself. This means that you can legally download the same operas you've already purchased, IMHO.
In other words you don't infringe anything if you download an MP3 of an original Vinyl that you own, for example...
Or, if in your country you have the fair use, you can also legally claim that you download for backup purposes.
Ok, but AFAIK "here in Italy" and anywhere in the world, there's no way to convince a legal on-line distributor, (which, by the way, may also reside in other countries and willing to apply his own law) that you're legally entitled to download a copy of a recording for free (plus at more a reasonable fee for distribution costs) because you already own a copy of it, maybe on a 20 years old vinyl!
And, to extend your reasoning, I've never tried to go to a shop and show my vinyl copy of a CD re-issue to have a reduction in price. My fault?
I thought you were talking about new purchases. I would have never asked had I thought you were simply stating the obvious. Thanks for further clarifying.
No problem, I should have stated that from the very get-go. I have no issues with new music being released in said lossy example (sourced from 24-bit files) and I would even embrace it if they offered free downloads of previously paid material. That being said, we know it will never happen just as me taking my Dad's old British Steel vinyl album into Best Buy and asking for the CD version for free isn't going to happen either. I would likely pay an extra fee (if Apple kept the $0.30 a track and $3.00 an album pricing) to upgrade to lossless tracks but not for lossy files again. I still haven't read anything concrete saying that they (along with others) are in talks of offering 24-bit lossless files though. The general wording I am seeing is "24-bit files" which could mean lossy or lossless files (though the labeling as 24-bit audio tends to make me think they are lossless files). We will see. All of this is probably a year or more off from ever seeing the light of day.
It makes me wonder what the competition will do as well. We all know Apple would likely offer ALAC files (hopefully DRM-free) if the record companies went the lossless route. Would Amazon offer WAV or FLAC? How about the Zune Marketplace, would Microsoft push WMA Lossless (does it even support 24-bit encoding)? Then, in my opinion, it wouldn't really matter which music store is chosen as they all offer files in the same quality and you could easily convert to whatever device you have. In fact, switching to 24-bit lossless files may end my support of CD all together. Not because I could potentially ABX the 24-bit lossless files from the 16-bit CDs (something of which I have never tested) but rather because I could obtain a lossless archive by never having to lift my fat butt from the comfort of my Lay-Z-Boy to obtain a lossless archive almost instantly. That, to me, is extremely appealing even if a slightly higher price is in place ($10 for the ~256kbps lossy files and then $12-$15 for the 24-bit lossless ones, that wouldn't be too far off from CD prices).
Regarding the other point, I can understand that you don't see the need in having lossy files sourced from 24-bit material. I simply question why you specifically said you didn't want lossy files sourced from 24-bit material. How would you know and why would you care?
My lack of enthusiasm for 24-bit sourced lossy files comes from the inevitable flood of advertising (especially since we all know Apple likely isn't the only one going to do this) and being required to buy the tracks all over again (whether at full price or not) in order to replace my current 16-bit sourced lossy files that I have already purchased from the iTunes Store. I wouldn't mind doing this if they jump to 24-bit lossless files as I would be paying more for a lossless archive. I don't want to pay more, again, for new lossy files (this would be the second time the iTunes Store started offering new lossy content). They might sound great, I don't know. I just don't want to have yet another lossy "upgrade" pushed on me require even more money. I purchased a handful of tracks back when the iTunes Store was offering 128kbps DRMed music (about $50 worth over the period of ~3 years) and they wanted $0.30 per track ($15 for music I have already purchased) when upgrading to the iTunes Plus standard (which is now what Apple offers music in). I just don't want to pay an additional sum of money for lossy music that I have already purchased. I know that an Apple rep isn't going to hold a gun to my head to make me upgrade but its the whole concept of buying lossy music again that I don't like.
...but while they'll be a few audiofools who do this (or will they - audiophiles don't
do lossy), and the usual Apple fanboys, I can't see many people paying money to upgrade a lossy file to a lossy file that someone told them sounded better,
if it doesn't.
There really would have to be some other reason to "upgrade".
It's got to be more about new purchases. And I suspect it's got to be about lossless. Not much point otherwise. Lossy "sourced from 24-bits" (same bitrate!) is not even an excuse to sell you a bigger iPod!
Cheers,
David.
There have been similar comments already, but in my opinion Apple is mainly doing this as a way to generate more ITunes Store revenue and also differentiate their music offerings from those eating away at their market share. In terms of improved audio quality, I think it's going to be 99.9% hype and essentially zero reality.
I do agree it would be nice if you can have 24 bit lossless files at no additional cost for archiving but how likely is that? I understand Apple is working on a cloud-based "backup" offering that will also let you share your Apple media wirelessly between devices.
Apple may have some higher-end tricks up their sleeve for playing media at home with new Apple TV improvements, etc. And this might also be part of the 24 bit strategy.
I plan to go buy one of the first 24 bit iPods and subject it to the dScope to see if there are measurable performance gains. There's a lot of hype and spin already around the "new Wolfson 24 bit DACs" they're supposedly going to be using.
I've already posted some results for the iPod Touch 3G on my blog. And with the exception of the 7 ohm output impedance, and perhaps being a bit weak on output power for some headphones, the DAC and audio performance is already excellent. So it's going to be hard to improve on the 16 bit performance. For those interested, I compare it to the Sansa Clip+ including jitter analysis, DAC linearity, etc:
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Very nicely done! Thanks for making the effort, its a breath of fresh air to see someone deal with data, and nice conversations around output impedance, etc.
Is this supposed "new" hardware gonna be needed for lossy conversions of these (hopefully) 24-bit lossless files? I think I understand why lossy codecs don't have bit depth like people said above, but when decoded, does this become 24-bit LPCM like the source was?
Very nicely done! Thanks for making the effort, its a breath of fresh air to see someone deal with data, and nice conversations around output impedance, etc.
Thanks! I'm still working on formatting/template issues with the blog. I just noticed the links in the tech section of that article all got changed to the font color so they were hard to notice. It's fixed now.
And yes, I'm trying to put the emphasis on the data. I also want to try and provide data that's more comparable (and repeatable/verifiable) than RMAA measurements as well as provide measurements you can't normally get with RMAA.
It will be very interesting to see how the 24 bit Apple hardware measures up. Perhaps if word gets around there's little actual advantage to the new DAC, it might prevent at least a few people from wasting their money on YAiPU (Yet Another i Pod Upgrade) in the quest of better audio quality. Of course the 5G Touch will probably have a camera in it and other compelling reasons to want one.
And speaking of output impedance... I've yet to test the iPhone 4 or a 4G Touch. But I've heard somewhat reliable information they have a much lower output impedance than my 3G. If that's true, the 4G Touch should be a near-perfect player in terms of audio performance as long as it will get loud enough without clipping for a given person's headphones and listening tastes--even without a Wolfson 24 bit DAC.
Is this supposed "new" hardware gonna be needed for lossy conversions of these (hopefully) 24-bit lossless files? I think I understand why lossy codecs don't have bit depth like people said above, but when decoded, does this become 24-bit LPCM like the source was?
At some point during playback, all compressed files--lossy or lossless--get turned back into regular PCM. Where and how that happens depends on what's doing the playback but generally that's all done in software.
Apple is famous for using either proprietary (or relatively uncommon) standards so they could invent their own flavor of 24 bit audio. And they like DRM (copy protection) and have been reluctant to fully get rid of it. So you could be stuck playing the new 24 bit format on only Apple hardware. It could be only the new 24 bit hardware (to help drive more hardware sales), or it could be anything you can upgrade to support the new format (Macs and iOS devices and possibly older iPods). Or, if they wanted to be nice for a change, they could use an industry open standard like 24 bit FLAC and then you could play your purchased music on lots of things.
Or if you buy a 24 bit track you might get an automatic license to the 16 bit version as well. So you could just use whichever version was needed for a particular player. Perhaps someone knows more about this but most of what I've seen is only rumor and guessing (like I'm doing here)?
In terms of raw audio files (ignoring Apple and their mostly closed world for a moment) a 16 bit DAC can play a 24 bit file by just ignoring the 8 least important bits. So that part is easy. There are basically two kinds of players. Those that decode everything in software (the CPU does all the work) or those that are sometimes used in portable players that do the decoding in dedicated hardware that may well be on the same chip as the DAC. These sorts of players can't ever support new formats.
Doing the decoding in software means you have total flexibility to handle new formats, etc--including 24 bit. Have a look at Rockbox (http://rockbox.org) for example. They add a bunch of new playback formats to dozens of different players by just changing the firmware.
But, if Apple wanted to have a more secure DRM scheme, they could do some of the decoding or DRM in hardware if they wanted--kind of like they have with some aspects of the iPad design. And that would make it much more difficult for anyone else to "hack" the 24 bit DRM audio files.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Very nicely done! Thanks for making the effort, its a breath of fresh air to see someone deal with data, and nice conversations around output impedance, etc.
Yes, that is great.
Though I'm becoming increasingly annoyed with my Clip+ and may try to find an old Clip on eBay: you can "lock" the Clip with the power off. You can't "lock" the Clip+ unless it's switched on. Which means... it gets switched on all by itself in my jacket pocket, and when I want to use it the battery is flat. Again. and again. and again. Oh, it crashes sometimes too.
(sorry for the OT post - but it's simple stuff like this that can make an otherwise wonderful player totally useless!)
Cheers,
David.
Though I'm becoming increasingly annoyed with my Clip+ and may try to find an old Clip on eBay: you can "lock" the Clip with the power off. You can't "lock" the Clip+ unless it's switched on. Which means... it gets switched on all by itself in my jacket pocket, and when I want to use it the battery is flat. Again. and again. and again. Oh, it crashes sometimes too.
I haven't encountered either problem, but I usually keep mine either clipped to me while using it, or in small pocket on my backpack when not (or laying around somewhere).
Are you running the Sansa or Rockbox firmware? If you're not running the latest Rockbox you should give it a try and it might not crash? I doubt Rockbox can change the power switch behavior, however. And if it's still crashing, report it to Rockbox. They do new builds nightly I think. But it might be your hardware? I didn't read all 1500 reviews on Amazon but I don't remember crashing being among the common complaints.
You could also always try the latest Sansa firmware as well if you haven't already. It's not under the support/firmware part of their website, but in their forum. But I've never had either one crash. My old e260 would crash fairly regularly--sometimes in really odd ways... like dropping one channel while the other kept playing with all the controls locked up.
Getting somewhat back on topic, I'm also likely to test the next generation of the Shuffle and/or Nano as well and report on those. But, for now, I'm really happy with the Clip+ (and its bigger brother the Fuze). And I'm looking forward to the Touch 5G mainly because I hope they fix the output impedance problem but also to see if the new 24 bit capability is pure marketing fluff.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Very nicely done! Thanks for making the effort, its a breath of fresh air to see someone deal with data, and nice conversations around output impedance, etc.
Yes, that is great.
Though I'm becoming increasingly annoyed with my Clip+ and may try to find an old Clip on eBay: you can "lock" the Clip with the power off. You can't "lock" the Clip+ unless it's switched on. Which means... it gets switched on all by itself in my jacket pocket, and when I want to use it the battery is flat. Again. and again. and again. Oh, it crashes sometimes too..
Never happens to me, but I don't use my clip a lot, mostly only camping.
What else do you carry in those pockets?
I've had serous problems with me destroying flash drives that I carried in my pocket until my wife bought me a flash drive cozy a couple of years back. I can't find a listening for the plain black one that she gaie me, but this is about as close as I hae found on the web:
Link to USB Flash Drie keychain cozy (http://www.etsy.com/listing/66676992/chapstickusb-cozy-key-chain-its-the-balm)
Problem went away. Maybe if you carry your Clip+ in a small bag like the one that comes with some earphones...
Apple is famous for using either proprietary (or relatively uncommon) standards so they could invent their own flavor of 24 bit audio. And they like DRM (copy protection) and have been reluctant to fully get rid of it. So you could be stuck playing the new 24 bit format on only Apple hardware. It could be only the new 24 bit hardware (to help drive more hardware sales), or it could be anything you can upgrade to support the new format (Macs and iOS devices and possibly older iPods). Or, if they wanted to be nice for a change, they could use an industry open standard like 24 bit FLAC and then you could play your purchased music on lots of things.
That's very baseless speculation. DRM was mandated by music labels. Apple pushed forward on February 6, 2007 to get rid of it. First there was some resistance by the big labels, as of April 2009, all music on the iTunes store is available DRM free. If you call AAC "uncommon", it is the official and advanced successor of MP3. Tech companies should not be criticized for
not being reactionary. ALAC already supports 24 bit, so speculation about a new "24 bit format" is without basis. While ALAC wasn't documented there are free implementations that work fine.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Most portables are used with IEMs. Since those are most of the time very sensitive, the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
Clip+ crashes on all Sansa firmware, including latest. Happens only when going into podcast list (I have lots in there). It's occasional (maybe once every five hours?). It reboots itself just fine.
Thanks for the suggestions for an external mechanical solution to prevent the Clip+ power control getting pushed. I suppose that would do. But the Clip had a perfect mechanical control built-in, and they removed it for the Clip+
Cheers,
David.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Most portables are used with IEMs. Since those are most of the time very sensitive, the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
I generally like the tests, but find the weight given to measurements made near FS to be excessive, for the reason given above. The AES allows measurements to be made at FS -3Db which I think is more representative of how equipment is used. I also think that measurements made at FS -10 dB are very relevant to actual use.
To me the most important thing about operation near FS is that the clipping is reasonably clean and lacks major nasties such as bursts of oscillation around 400 Hz, which I have seen at times. The possiblity that a digital player or DAC is going to see signals above -10 dB @ > 15 KHz is IME about zero. Reality is more like -40 dB. Oh, I'm sure that there is one second of some track some place that violates this, but...
Apple is famous for using either proprietary (or relatively uncommon) standards so they could invent their own flavor of 24 bit audio. And they like DRM (copy protection) and have been reluctant to fully get rid of it. So you could be stuck playing the new 24 bit format on only Apple hardware. It could be only the new 24 bit hardware (to help drive more hardware sales), or it could be anything you can upgrade to support the new format (Macs and iOS devices and possibly older iPods). Or, if they wanted to be nice for a change, they could use an industry open standard like 24 bit FLAC and then you could play your purchased music on lots of things.
That's very baseless speculation. DRM was mandated by music labels. Apple pushed forward on February 6, 2007 to get rid of it. First there was some resistance by the big labels, as of April 2009, all music on the iTunes store is available DRM free. If you call AAC "uncommon", it is the official and advanced successor of MP3. Tech companies should not be criticized for not being reactionary. ALAC already supports 24 bit, so speculation about a new "24 bit format" is without basis. While ALAC wasn't documented there are free implementations that work fine.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Most portables are used with IEMs. Since those are most of the time very sensitive, the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
Well the DRM topic is open to debate, but other companies somehow were able to drop DRM before Apple did with some of the same music lablels. And it's generally well accepted that Apple tends to do things in a way that keep you buying other Apple products. A good example is the video chat with the iPhone 4. There were widely accepted open standards they *could* have used, but they chose a proprietary closed solution instead that only works to other iPhone users. The same is also true of some of their Bluetooth implementations--they only support Apple authorized devices. I could keep going, but you get the idea.
And yes, I was referring to inventing ALAC when there were perfectly good open lossless standards already in place. But yes, a lot of what I posted is speculation. I admitted as much.
As for your IEM comment relative to my testing, I agree 100%. I wrote an entire blog article on why RMAA testing often doesn't work well because there are no absolute levels, among other problems. That's why I do most of my testing with a $10,000 Prism Sound dScope using a variety of loads including IEMs. The dScope is vastly superior to *ANY* soundcard that I'm aware of and I explain why in the links below:
RMAA Testing (http://nwavguy.blogspot.com/2011/02/rightmark-audio-analyzer-rmaa.html)
Testing Methods (http://nwavguy.blogspot.com/2011/02/testing-methods.html)
A good example is the video chat with the iPhone 4. There were widely accepted open standards they *could* have used, but they chose a proprietary closed solution instead that only works to other iPhone users.
Which would be?
A good example is the video chat with the iPhone 4. There were widely accepted open standards they *could* have used, but they chose a proprietary closed solution instead that only works to other iPhone users.
Which would be?
I was thinking of DVB-H and Nokia's video calling (which predates Apple's by several years) but, in looking into it further, they're not (yet at least) fully interoperable either. Aylus does have an interoperable standard, but it's newer than the iPhone 4.
So I rephrase my comment to "so far Apple's FaceTime is a Apple-only solution". But we're getting further and further OT here.
I don't think it's too far fetched that DRM will come back with lossless music though. But also I don't doubt outrage from many consumers.
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Most portables are used with IEMs. Since those are most of the time very sensitive, the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
I don't understand what you're getting at here. "a sound card with matching input sensitivity" would mean something like a preamp right? Why is one of those needed to get meaningful results?
Sansa Clip+ vs iPod Touch 3G Detailed Measurements (http://nwavguy.blogspot.com/2011/02/sansa-clip-measured.html)
Most portables are used with IEMs. Since those are most of the time very sensitive, the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
I don't understand what you're getting at here. "a sound card with matching input sensitivity" would mean something like a preamp right? Why is one of those needed to get meaningful results?
I've been testing all sorts of things with RMAA, and did similar things with Spectra Lab before that. FWIW many of the screens and menus shown on the NwAvGuy's web site (apparently screen shots from the Prism test set he loves so dearly) look very much to me like Spectra Lab.
I see many of his complaints with RMAA as being procedural things. For example he says:
"RMAA has no concept of absolute levels. So it can't measure voltages, power outputs, etc. So you have no way of knowing, for example, how loud that portable player or headphone amp can play without obvious distortion"
The real problem is not RMAA, its the fact that audio interfaces aren't formally calibrated. To me this is just a matter of common sense - if you want to know the voltage, get a voltmeter! Ihave a Fluke 85 that has very flat frequency response to over 100 KHz.
When I use RMAA (or Spectra) to test equipment whose voltage is outside of the reasonable realm of use of the audio interface, I add an external amplifier or attenuator.
It turns out that a high quality mic preamp makes an flexible and effective gain block for measuring low voltage signals. When used at reasonable gains in the 0-20 dB range they are typically very clean.
I use RMAA to test power amps by means of a high-powered dummy load (8 pieces 8 ohm 300 watt non-inductive resistors with some switches, connectors and a fan) that has a built-in attenuator. I also have a loudspeaker simulator with similar features and capabilities.
If I want low-residual measurements, I use a really good audio interface. For example the M-Audio Delta AP 24192 has residuals that are about 110 dB down and costs less than $200. This gives you a goodly amount of wiggle room if you want to use it to measure signals that aren't near FS at its input.
High frequency IM tests can be done with RMAA by simply setting the IM test frequencies to 19 and 20 KHz. The test frequencies can be reset and jitter can be analyzed from the spectrum analysis plots.
Etc., etc.
I don't understand what you're getting at here. "a sound card with matching input sensitivity" would mean something like a preamp right? Why is one of those needed to get meaningful results?
A typical IEM already outputs far over 100 dB of sound pressure (106 dB in my case) for 1 millivolt input voltage. At these low levels output performance may differ considerably from the kind of measurement one usually gets from a high impedance line-level input. No, loading with a parallel resistor doesn't help.
Since this is OT, we could continue the discussion [a href='index.php?act=findpost&pid=86306']here[/a].
A typical IEM already outputs far over 100 dB of sound pressure (106 dB in my case) for 1 millivolt input voltage. At these low levels output performance may differ considerably from the kind of measurement one usually gets from a high impedance line-level input.
Ok but you said:
the RMAA measurements don't reflect actual performance. You'd need a sound card with matching input sensitivity to get meaningful results.
It sounded like you had results or a concrete theory here, not just speculation. Is that the case?
It sounded like you had results or a concrete theory here, not just speculation. Is that the case?
I have posted my findings with the Sansa Clip+ in the linked thread. They were independently confirmed. RMAA results are very good, but the described noise is clearly audible, I'd say at max -50 dB down. When I try to record it into the high-impedance line-in of my sound card there is at least -90 dB of silence.
It sounded like you had results or a concrete theory here, not just speculation. Is that the case?
I have posted my findings with the Sansa Clip+ in the linked thread.
Which thread is this? I'd love to see the results.
Sorry, the link was broken. This is correct:
Since this is OT, we could continue the discussion [a href='index.php?showtopic=86306']here[/a].
I've been testing all sorts of things with RMAA, and did similar things with Spectra Lab before that. FWIW many of the screens and menus shown on the NwAvGuy's web site (apparently screen shots from the Prism test set he loves so dearly) look very much to me like Spectra Lab.
I see many of his complaints with RMAA as being procedural things. For example he says:
"RMAA has no concept of absolute levels. So it can't measure voltages, power outputs, etc. So you have no way of knowing, for example, how loud that portable player or headphone amp can play without obvious distortion"
The real problem is not RMAA, its the fact that audio interfaces aren't formally calibrated. To me this is just a matter of common sense - if you want to know the voltage, get a voltmeter! Ihave a Fluke 85 that has very flat frequency response to over 100 KHz.
When I use RMAA (or Spectra) to test equipment whose voltage is outside of the reasonable realm of use of the audio interface, I add an external amplifier or attenuator.
It turns out that a high quality mic preamp makes an flexible and effective gain block for measuring low voltage signals. When used at reasonable gains in the 0-20 dB range they are typically very clean.
I use RMAA to test power amps by means of a high-powered dummy load (8 pieces 8 ohm 300 watt non-inductive resistors with some switches, connectors and a fan) that has a built-in attenuator. I also have a loudspeaker simulator with similar features and capabilities.
If I want low-residual measurements, I use a really good audio interface. For example the M-Audio Delta AP 24192 has residuals that are about 110 dB down and costs less than $200. This gives you a goodly amount of wiggle room if you want to use it to measure signals that aren't near FS at its input.
High frequency IM tests can be done with RMAA by simply setting the IM test frequencies to 19 and 20 KHz. The test frequencies can be reset and jitter can be analyzed from the spectrum analysis plots.
I say in the first paragraph of my article RMAA is certainly useful when it's used correctly and people understand the limitations. But a lot of people using RMAA don't have the tools, skills or knowledge to use it correctly. Yet they still post their results with no disclaimer indicating they were taken at unknown levels, often with an unknown load, unknown sound hardware, etc. So for the Average Joe using Google to find test results on some piece of equipment, the odds actually favor finding poorly done RMAA results.
People like you are the exception among RMAA users, not the rule.
You're correct some issues are procedural, and at the end of the article I talk about, for example, using a good DMM like your Fluke, to set levels. And, with enough effort, knowledge, and a good variety of external hardware, voltage dividers, other software programs, etc. you can extract a decent set of results from a good soundcard. But most don't do all that correctly, or even at all, with RMAA.
And, even in the best of hands, RMAA still falls flat in several other areas. It's rather fixed in what it does, and AFAIK, it's not well documented how it's doing some things internally. This makes comparing RMAA results, to results obtained other ways, more difficult (or some would argue impossible). With an instrument-grade analyzer like an Audio Precision or dScope you have total control over most everything right down to the amount of jitter and sample rate deviation you want in the digital output signal for testing a DAC's ability to handle less than ideal digital inputs. You can configure the inputs and outputs just about any way imaginable with any combination of digital/analog/soundcard generated signals and similar on the analysis side.
The dScope measures the level at its inputs from microvolts to 159 volts with 0.05 dB accuracy without needing any dividers, meters, etc. It has seamless autoranging that keeps the A/D running within 2 dB of it's 0 dBFS level. Try THAT with a soundcard! External dividers are typically far more "coarse" and can be a source of considerable error depending on resistor tolerances, sound card input impedance, device source impedance, etc.
It's incredibly useful to have digital and analog outputs, and digital and analog inputs, that are
always referenced to known levels and each other. There's no need to constantly use a meter to set levels on both sides. And, of course, you can't "meter" a digital signal at all. But dScope always knows what dBu, dBv, etc. are. They're not just relative abstract numbers like they are with a soundcard. And it can relate those numbers on both sides to dBFS in the digital domain which is really useful.
And sound cards with analog level controls (like say the popular E-Mu 0202, 0404, Pre, M-Audio, etc.) make it difficult to change the levels without also messing up the channel balance. So you need to get your meter out every time you even bump those controls. And the controls on the 0202 and 0404 cover such a wide range (65 dB I think?) they sometimes change themselves they're so touchy. They're the opposite of being accurate.
And sound cards that use digital "mixer based" controls typically have fixed steps that might be too coarse to say do something like correct for a 0.7 dB channel imbalance. You're stuck with 1 dB or 2 dB steps for each channel. And many are not calibrated in dB at all. They're "mystery steps".
The dScope has precise selectable input impedance settings which sound cards lack. And perhaps most important, it has floating differential input and outputs to avoid ground loops, or worse, that can be a serious problem trying to use a typical soundcard. Some power amps will self destruct if you ground their output "grounds" to real ground like most sound cards do. And ground loops can easily contaminate sound card measurements.
And the dScope (unlike the Audio Control AP5xx) has what they call a Continuous Time Analyzer. This lets you do true real time analysis without needing to first run FFT passes like RMAA and Spectra Lab do. This can be incredibly useful during testing in many different ways--especially when combined with the scripting built into the dScope software. The dScope can, for example, regulate the input signal driving an amplifier to maintain a constant THD level at the output of the amplifer while doing a live frequency sweep so you end up with a very useful power at xxx % THD vs frequency plot.
The dScope also has, built in, a wide variety of industry standard measurement filters and test signals. Like CCIR, A Weighting, C Weighting, SMPTE, etc. And it can implement those filters in real time rather than applying them after the fact to a completed FFT. You can also listen to the live distortion portion, or even just the Nth harmonic, via the monitor output live if you want. I could keep going, but hopefully you get the idea.
To put it another way, the people designing the soundcards, or any other piece of respectable audio gear, you admire are very likely not using a soundcard to design and measure them. Odds are they're using an Audio Precision or dScope analyzer.
It sounded like you had results or a concrete theory here, not just speculation. Is that the case?
I have posted my findings with the Sansa Clip+ in the linked thread. They were independently confirmed. RMAA results are very good, but the described noise is clearly audible, I'd say at max -50 dB down. When I try to record it into the high-impedance line-in of my sound card there is at least -90 dB of silence.
If I understand you correctly, you comparing an unloaded RMAA test to a loaded RMAA test? I think its expected that the loaded test will be much worse, which is why people generally only compare loaded RMAA results. If I understand what you are suggesting, I think you would have to use a good preamp, a parallel load, and then test at different volume settings.
And, even in the best of hands, RMAA still falls flat in several other areas. It's rather fixed in what it does, and AFAIK, it's not well documented how it's doing some things internally. This makes comparing RMAA results, to results obtained other ways, more difficult (or some would argue impossible). With an instrument-grade analyzer like an Audio Precision or dScope you have total control over most everything right down to the amount of jitter and sample rate deviation you want in the digital output signal for testing a DAC's ability to handle less than ideal digital inputs. You can configure the inputs and outputs just about any way imaginable with any combination of digital/analog/soundcard generated signals and similar on the analysis side.
While I always like to see better testing performed, reading your results I'm not sure I see anything being revealed that we didn't already know from RMAA? You've certainly measured parameters more accurately then previous tests, but the more accurate results don't really seem to change anything we thought we knew from RMAA.
Which makes it a little odd that you're so concerned about people using RMAA to compare things. If anything, your own results seem to have confirmed that its a good enough tool for what people want to do with it. Or am I missing something?
[To put it another way, the people designing the soundcards, or any other piece of respectable audio gear, you admire are very likely not using a soundcard to design and measure them. Odds are they're using an Audio Precision or dScope analyzer.
Interesting factoid: I happen to own an Audio Precision System 1 test set that I hardly touch. I kinda sorta inherited it from Julian Hirsch.
Relevent fact: Last time I looked the DScope was about $10 grand, while the most expensive piece of equipment I've described so far is under $200.
Here's how I approach the issue of test equipment - one of my goals in life is to encourage people to equip themselves to better understand how audio works. I think that hands on experience with hardware is very helpful. By sticking with budget approaches I keep myself familiar with the slings and arrows of going down that road. And on occasion
I've published auiod equipment test reports that it is said proved performance in excess of that possible with the best equipment then available.
While I always like to see better testing performed, reading your results I'm not sure I see anything being revealed that we didn't already know from RMAA? You've certainly measured parameters more accurately then previous tests, but the more accurate results don't really seem to change anything we thought we knew from RMAA.
Which makes it a little odd that you're so concerned about people using RMAA to compare things. If anything, your own results seem to have confirmed that its a good enough tool for what people want to do with it. Or am I missing something?
I partly agree with you. But there are some important things that RMAA can't directly measure, such as the maximum output level and output impedance, which are critical measurements when comparing portable players. It also can't measure the famous pitch error of the Sansa. Yeah, you may have known about all 3 of those numbers, but they don't show up in any RMAA table, graph or report. It also doesn't measure the DAC linearity (unless you count the -60 dB spectrum plot but virtually any DAC has near zero error there) or square wave response. And remember I was using a really high-end $1800 ADC with RMAA to make those measurements. And I was doing it at known levels, with known loads, etc. Others may not get nearly as similar results with more typical RMAA usage.
Because I picked one of the more popular (non Apple) players on the planet, there's no shortage of test data for it. It also helps that it's a Rockbox player as that means a higher percentage of serious geeks are involved. But if you look at a niche product, like the Cowon i9, it's a lot harder to scare up valid numbers. And the sort of "peer review" process the Clip+ results have already been subjected to is often totally missing. So you may well just get a single set of RMAA numbers and graphs with little clue if you can trust them or not. And, for the reasons I've outlined, I'd bet on "not" most of the time.
Interesting factoid: I happen to own an Audio Precision System 1 test set that I hardly touch. I kinda sorta inherited it from Julian Hirsch.
That's kinda fun! We're getting way OT here, but the AP System 1 is the beast that started it all, and so in some ways, was Julian Hirsch. Both, in their own way, are legends.
If I had a System 1 I wouldn't use it either except perhaps as ballast. It's big, heavy, and uses a proprietary interface that originally only worked with a compatible ancient 8 bit ISA slot (aka as found in the original IBM XT). And the software is DOS based. I'm not sure what AP offers today in terms of interfaces, but it's probably not worth the expense to buy a modern interface for it if one is even available.
It also doesn't measure the DAC linearity (unless you count the -60 dB spectrum plot but virtually any DAC has near zero error there) or square wave response.
Sigma Delta DACs (the industry standard for about the past decade or more) can't possibly have linearity problems. That's one reason why virtually every DAC has low error at FS-60 dB .
Of course if every DAC actually had near zero error at -60 dB then they would all give the same results in a standard dynamic range test. But they don't.
As far as square wave response goes, this sort of thing should really be done with an ADC that is sampling at a fir higher rate then the UUT or an actual oscilloscope.
Square wave testing has gone out of favor because any related sonic problems would show up in a frequency response test.
I partly agree with you. But there are some important things that RMAA can't directly measure, such as the maximum output level and output impedance, which are critical measurements when comparing portable players.
But are those really so useful to know? Pretty much every player in existence has an output level with 3-4 dB of every other player, since they're all running off the same supply voltage. Yeah its interesting to know if your player is marginally louder then another player, but theres so little difference between players I'd hardly consider it critical. Likewise, if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?
It also can't measure the famous pitch error of the Sansa.
Thats a fair point, although this is a pretty rare problem.
Yeah, you may have known about all 3 of those numbers, but they don't show up in any RMAA table, graph or report. It also doesn't measure the DAC linearity (unless you count the -60 dB spectrum plot but virtually any DAC has near zero error there) or square wave response.
Wouldn't DAC nonlinearity show up in the distortion tests though? Yeah you won't get the exact number, but do you really need that?
And remember I was using a really high-end $1800 ADC with RMAA to make those measurements. And I was doing it at known levels, with known loads, etc. Others may not get nearly as similar results with more typical RMAA usage.
Yeah, but heres the thing: the stuff people actually want to measure doesn't require a good ADC like that. A good onboard sound card will tell you if the frequency response is flat from 100Hz to 16kHz, if the noise floor is unreasonably high, and if theres a lot of distortion. If I think about audio quality at all when buying a portable player, those are the only three numbers I'd really care about.
But are those really so useful to know? Pretty much every player in existence has an output level with 3-4 dB of every other player, since they're all running off the same supply voltage. Yeah its interesting to know if your player is marginally louder then another player, but theres so little difference between players I'd hardly consider it critical. Likewise, if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?
Yes they are important! The load interacts with the output impedance in a large ways that have little to do with the power supply voltage. And some amps simply don't have the ability to drive low impedance loads to typical levels before the distortion goes off the scale. For example, most portable headphones these days are 16 ohms. And into 15 ohm loads, I've measured anywhere from 179 mV RMS at 1 % THD up to 800+ mV. That's a range of 2.1 mW up to 42.7 mW--a 13 dB difference not a 3-4 dB difference as you suggest.
For example, here's what happens with a 6 ohm output impedance device into typical balanced armature (Ultimate Ears SuperFi in this case) headphones:
(http://lh5.ggpht.com/_gnU30vMbtYg/TW2agRwyM1I/AAAAAAAAARA/Y2Lmw4TbwW0/NuForce%20uDAC-2%20-3%20dBFS%20Swept%20Frequency%2015%20Ohms%20(yellow)%20UE%20SuperFi%205's%20(blue)%20(Ref%20~400%20mV)_thumb[2].png)
Does 4+ dB of variation matter? Some devices have 20 - 200 ohm output impedance--far greater than the 6 ohms show above. And, obviously, cause far greater deviations from flat frequency response.
Wouldn't DAC nonlinearity show up in the distortion tests though? Yeah you won't get the exact number, but do you really need that?
That's a fair question, and based on what I know and have seen, the answer is: "not usually". It's really tricky to take accurate distortion measurements at very low levels as many of the harmonics are lost in the noise. And if it's THD+N the entire measurement is dominated by random noise, not the non-linearity error. And when you test at higher levels, other sources of distortion usually dominate the measurement. So DAC nonlinearity can be lost either way.
Yeah, but heres the thing: the stuff people actually want to measure doesn't require a good ADC like that. A good onboard sound card will tell you if the frequency response is flat from 100Hz to 16kHz, if the noise floor is unreasonably high, and if theres a lot of distortion. If I think about audio quality at all when buying a portable player, those are the only three numbers I'd really care about.
So far I've been concentrating on portable equipment, and for that, I mostly agree with you. But I don't have a dScope just to test $29 portable players. That said, even the iPod Touch 3G will give many sound cards a good run for their money in terms of noise and distortion. So it's useful to get real numbers and not be flirting with the limitations of the PC's sound hardware.
It's also worth pointing out there are all sorts of other measurements I could be showing results for that I haven't (yet). They include linearity plots, digital deviation and jitter (for digital interfaces), the output power vs THD, various phase measurements, impulse responses, etc. My reviews are already long enough for now.
So the bigger issue is all the things that RMAA and soundcards don't measure directly at all. And part of it is convenience. If it takes 3 days to run a full battery of tests using a kludged together collection of "tools" you're a lot less likely to ever do all the tests in the first place. If it's a matter of connecting the device, setting a calibration level in the dScope software, and clicking the "GO" button to run the script, that's very different.
RMAA is great in that it's easy to use. But, as I've explained, it's very much "garbage in/garbage out" and you're still missing lots of important things (including what we just discussed above). To get the rest of the numbers with a soundcard, meter, etc. is considerably more work. And even with all that there are still typical specs you can't measure (or measure very well). Like I said, RMAA and soundcards are certainly useful. But I don't think it's accurate to imply an audio analyzer has little advantage making real world measurements.
But are those really so useful to know? Pretty much every player in existence has an output level with 3-4 dB of every other player, since they're all running off the same supply voltage. Yeah its interesting to know if your player is marginally louder then another player, but theres so little difference between players I'd hardly consider it critical. Likewise, if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?
Yes they are important! The load interacts with the output impedance in a large ways that have little to do with the power supply voltage. And some amps simply don't have the ability to drive low impedance loads to typical levels before the distortion goes off the scale. For example, most portable headphones these days are 16 ohms. And into 15 ohm loads, I've measured anywhere from 179 mV RMS at 1 % THD up to 800+ mV. That's a range of 2.1 mW up to 42.7 mW--a 13 dB difference not a 3-4 dB difference as you suggest.
For example, here's what happens with a 6 ohm output impedance device into typical balanced armature (Ultimate Ears SuperFi in this case) headphones:
(img)
Does 4+ dB of variation matter? Some devices have 20 - 200 ohm output impedance--far greater than the 6 ohms show above. And, obviously, cause far greater deviations from flat frequency response.
Note that I said, "if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?". I'm aware that lots of devices don't have a perfectly flat spectrum when driving low impedances. But if RMAA tells you one does, what additional benefit does output impedance get you exactly?
Furthermore, I don't know what you mean by "some devices have 20 - 200 ohm output impedance". Certainly not digital audio players, as they would waste so much battery power just heating their output buffering stage.
Wouldn't DAC nonlinearity show up in the distortion tests though? Yeah you won't get the exact number, but do you really need that?
That's a fair question, and based on what I know and have seen, the answer is: "not usually". It's really tricky to take accurate distortion measurements at very low levels as many of the harmonics are lost in the noise.
Does that matter? If a DAC is nonlinear at low levels, and you measure nonlinear mixing between two near full scale sin waves, then you will get a fairly large amplitude peak at sum or difference frequencies. I don't think you actually have to generate a low intensity signal to measure this effect.
And when you test at higher levels, other sources of distortion usually dominate the measurement. So DAC nonlinearity can be lost either way.
Err, if distortion is already generally negligible on most DACs, and this effect is so small that its masked by other distortion, then what purpose does measuring it serve? What you actually want to check for is audible levels of nonlinearity. If nonlinearity is so low you cannot practically measure it, then it is clearly inaudible.
So far I've been concentrating on portable equipment, and for that, I mostly agree with you. But I don't have a dScope just to test $29 portable players. That said, even the iPod Touch 3G will give many sound cards a good run for their money in terms of noise and distortion. So it's useful to get real numbers and not be flirting with the limitations of the PC's sound hardware.
Yes, but you specifically complained about people using RMAA to compare MP3 players. I'm saying that this is unfair, its a perfectly adequate tool for doing so.
RMAA is great in that it's easy to use. But, as I've explained, it's very much "garbage in/garbage out" and you're still missing lots of important things (including what we just discussed above). To get the rest of the numbers with a soundcard, meter, etc. is considerably more work. And even with all that there are still typical specs you can't measure (or measure very well). Like I said, RMAA and soundcards are certainly useful. But I don't think it's accurate to imply an audio analyzer has little advantage making real world measurements.
I don't think you've explained how its "garbage in/garbage out" nor how its "missing lots of important things". You've listed things it missing, but you haven't made a convincing argument that any of them are important.
It also doesn't measure the DAC linearity (unless you count the -60 dB spectrum plot but virtually any DAC has near zero error there) or square wave response.
Sigma Delta DACs (the industry standard for about the past decade or more) can't possibly have linearity problems. That's one reason why virtually every DAC has low error at FS-60 dB .
Of course if every DAC actually had near zero error at -60 dB then they would all give the same results in a standard dynamic range test. But they don't.
As far as square wave response goes, this sort of thing should really be done with an ADC that is sampling at a fir higher rate then the UUT or an actual oscilloscope.
Square wave testing has gone out of favor because any related sonic problems would show up in a frequency response test.
I should probably give it up, but there are plenty of examples of Delta-Sigma DACs with linearity error. It can be caused by lots of things including the power supply/voltage reference used in the implementation or noise sources. If you'd like, I'll point you to some references measured with Audio Precision gear. I've also seen it personally.
And why do you care about square wave information on an audio device that's beyond 96 Khz? That's the only difference between viewing the square wave of a faster scope vs the dScope. In fact, because most faster scopes have only 8 bit DACs, once can argue the 24 bit DACs in the dScope actually yield more useful information--especially if you want to zoom in on a particular vertical anomaly (like say ringing, etc.). I have lots faster scopes (including a high-end 4 Gs/sec Agilent) but I don't use it for audio use when the dScope does a better job.
Square waves can tell you a lot about any amplifiers output stability (including feedback loop design) and also the type of digital/analog low pass (reconstruction) filtering used after a DAC--i.e. is it a linear phase design, etc. Neither of these show up in any consistent or conclusive way in frequency response tests.
Does that matter? If a DAC is nonlinear at low levels, and you measure nonlinear mixing between two near full scale sin waves, then you will get a fairly large amplitude peak at sum or difference frequencies. I don't think you actually have to generate a low intensity signal to measure this effect.
Some noise levels don't seem to scale linearly with output power, even when the DAC chip itself has good linearity, for example due to improper shielding/grounding/power supply/etc, and they are much more audible at lower levels. The measurement of those artifacts mostly relates to the recording device's (or headphone's) sensitivity and not to the loaded impedance. So you can get cases where RMAA looks excellent while your IEMs and better measurement equipment say otherwise.
Note that I said, "if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?". I'm aware that lots of devices don't have a perfectly flat spectrum when driving low impedances. But if RMAA tells you one does, what additional benefit does output impedance get you exactly?
If you're talking about
one person using their current headphones then yes, you can verify the device is sufficiently flat with RMAA. But what good is that information for all the other people trying to use the posted RMAA data that have different headphones? It's especially useless unless the person also posted an impedance plot of the headphones--which by the way is yet something else RMAA won't do that the dScope does. So what you suggest is really useless in terms of a general result that others can fairly use for comparison.
Furthermore, I don't know what you mean by "some devices have 20 - 200 ohm output impedance". Certainly not digital audio players, as they would waste so much battery power just heating their output buffering stage.
You would think that would be true, but it's not. I've measured output impedance of ~50 ohms on portable gear and my HP PC's headphone jack is 170 ohms. The iPod Touch 3G is 7 ohms.
Does that matter? If a DAC is nonlinear at low levels, and you measure nonlinear mixing between two near full scale sin waves, then you will get a fairly large amplitude peak at sum or difference frequencies. I don't think you actually have to generate a low intensity signal to measure this effect.
I guess the short answer is it matters to
some people. We can turn this into the endless complex discussion of what levels and kinds of distortions are audible, but that's not what you asked. I thought the issue was if DAC non-linearity was accurately reflected in THD measurements. The answer is no. If you want to ask if measurable DAC non-linearity
matters, that's a different question.
Yes, but you specifically complained about people using RMAA to compare MP3 players. I'm saying that this is unfair, its a perfectly adequate tool for doing so.
I maintain it's far from "unfair". I can point you to all sorts of misleading RMAA results on the web of portable players. A perfect example is I have an older Sansa player (or the new Cowon i9 for that matter) that rolls off in audible ways at bass frequencies when driving real 16 ohm headphones. But with no load, they're both flat to 20 hz. Many (most?) RMAA tests are run with no load but that's never disclosed. So it's easy to have misleading information.
And you're completely ignoring the 13 dB of power output difference not measured by RMAA. Or the fact that just because it shows reasonably flat response with dynamic headphones, the results could be wildly different with balanced armature headphones.
I don't think you've explained how its "garbage in/garbage out" nor how its "missing lots of important things". You've listed things it missing, but you haven't made a convincing argument that any of them are important.
That's your opinion. There are several examples of "garbage" in my RMAA blog article and I've presented several in this thread. If you're not convinced, that's fine. I guess we have very different ideas of what matters.
Some noise levels don't seem to scale linearly with output power, even when the DAC chip itself has good linearity, for example due to improper shielding/grounding/power supply/etc, and they are much more audible at lower levels. The measurement of those artifacts mostly relates to the recording device's (or headphone's) sensitivity and not to the loaded impedance. So you can get cases where RMAA looks excellent while your IEMs and better measurement equipment say otherwise.
You claimed this before and never answered back as to if you had tested (your Sansa) again with a load.
I believe you will see the noise in question if you perform the RMAA test loaded.
Does that matter? If a DAC is nonlinear at low levels, and you measure nonlinear mixing between two near full scale sin waves, then you will get a fairly large amplitude peak at sum or difference frequencies. I don't think you actually have to generate a low intensity signal to measure this effect.
Some noise levels don't seem to scale linearly with output power, even when the DAC chip itself has good linearity, for example due to improper shielding/grounding/power supply/etc, and they are much more audible at lower levels. The measurement of those artifacts mostly relates to the recording device's (or headphone's) sensitivity and not to the loaded impedance. So you can get cases where RMAA looks excellent while your IEMs and better measurement equipment say otherwise.
I'm not really sure how to interpret this response to my statement. Are you disagreeing with me about nonlinear mixing, or did you just quote the wrong block of text?
Note that I said, "if a player a sufficiently flat frequency response, is it really that important to know the precise output impedance?". I'm aware that lots of devices don't have a perfectly flat spectrum when driving low impedances. But if RMAA tells you one does, what additional benefit does output impedance get you exactly?
If you're talking about one person using their current headphones then yes, you can verify the device is sufficiently flat with RMAA. But what good is that information for all the other people trying to use the posted RMAA data that have different headphones? It's especially useless unless the person also posted an impedance plot of the headphones--which by the way is yet something else RMAA won't do that the dScope does. So what you suggest is really useless in terms of a general result that others can fairly use for comparison.
Its not useless at all. While different headphones have different impedance vs. frequency curves, in practice two people both using 16 ohm IEMs will typically get similar results unless one has particularly weird headphones. Thats why knowing the output impedance doesn't really get you anything.
Furthermore, I don't know what you mean by "some devices have 20 - 200 ohm output impedance". Certainly not digital audio players, as they would waste so much battery power just heating their output buffering stage.
You would think that would be true, but it's not. I've measured output impedance of ~50 ohms on portable gear and my HP PC's headphone jack is 170 ohms. The iPod Touch 3G is 7 ohms.
While I'm skeptical about those numbers, I'd like to point out that your PC is not a portable audio player, and so clearly not what I was referring to.
Does that matter? If a DAC is nonlinear at low levels, and you measure nonlinear mixing between two near full scale sin waves, then you will get a fairly large amplitude peak at sum or difference frequencies. I don't think you actually have to generate a low intensity signal to measure this effect.
I guess the short answer is it matters to some people. We can turn this into the endless complex discussion of what levels and kinds of distortions are audible, but that's not what you asked. I thought the issue was if DAC non-linearity was accurately reflected in THD measurements. The answer is no. If you want to ask if measurable DAC non-linearity matters, that's a different question.
Wait what? I'm saying that if your DAC has significant nonlinearity, that will be reflected in THD and IMD measurements. Thats why nonlinearity matters, it generates distortion. If its not generating distortion, then why would you care about it?
Yes, but you specifically complained about people using RMAA to compare MP3 players. I'm saying that this is unfair, its a perfectly adequate tool for doing so.
I maintain it's far from "unfair". I can point you to all sorts of misleading RMAA results on the web of portable players. A perfect example is I have an older Sansa player (or the new Cowon i9 for that matter) that rolls off in audible ways at bass frequencies when driving real 16 ohm headphones. But with no load, they're both flat to 20 hz. Many (most?) RMAA tests are run with no load but that's never disclosed. So it's easy to have misleading information.
Yes, if you test without a load the results aren't very relevant. But so what? Test with a load and you're good.
And you're completely ignoring the 13 dB of power output difference not measured by RMAA. Or the fact that just because it shows reasonably flat response with dynamic headphones, the results could be wildly different with balanced armature headphones.
Which two players have a 13 dB difference? And I don't doubt that some headphones exist that are extremely hard to drive, but they're the exception, not the norm.
That's your opinion. There are several examples of "garbage" in my RMAA blog article and I've presented several in this thread. If you're not convinced, that's fine. I guess we have very different ideas of what matters.
The only points I recall you making on your blog were complaints about people doing unloaded RMAA tests, and something about one of your tests rolling off high frequencies inexplicably which I think was actually do to some oddness with your sound card. Pardon me for finding these unconvincing. To convince me you would have to have some measurement that tells me something I didn't already know just from RMAA. Something surprising and useful, not merely something interesting in an academic sense.
While different headphones have different impedance vs. frequency curves, in practice two people both using 16 ohm IEMs will typically get similar results unless one has particularly weird headphones. Thats why knowing the output impedance doesn't really get you anything.
Knowing the output impedance gets you a lot. A device with a < 1 ohm output impedance will deliver it's measured frequency response within a fraction of a dB to nearly any headphone load. As the graph I posted earlier shows, one with even a 6 ohm output impedance clearly will not.
You don't find that useful? And certainly not everyone uses 16 ohm headphones. And even among those rated at 16 ohms, the differences can run wide. My SuperFi IEM's peak at 80 ohms and dip down to below 10 ohms and peak at about 1200 hz:
(http://1.bp.blogspot.com/_gnU30vMbtYg/TVLyRxiMOzI/AAAAAAAAABY/vEjVb4Ytm5w/s400/UE%20superfi5%20free%20air%20impedance%20phase.jpg)
And here's a pair of Sony IEMs which fluctuate by
only 1 ohm from 17 to 18 ohms and peak at 5000 hz:
(http://2.bp.blogspot.com/_gnU30vMbtYg/TVLrIH7cTFI/AAAAAAAAABU/mL0F1MaRYys/s400/Sony%20MDR-EX51%20impedance.jpg)
My Eytmotics have a totally different shaped curve than either of the above and peak at 100 ohms. Some IEM's have multiple resonances while double and triple driver balanced armature IEMs get even more weird. You apparently haven't measured, or seen, many of the impedance plots?
Can you honestly tell me the two headphones above, will yield "similar results" connected to the 7 ohm output impedance of an iPod Touch 3G? They won't. Not even close.
But they will connected to a Sansa Clip+. So your statement above, and the one earlier, are simply wrong. It does matter--at least to lots of people. If you don't care about audible frequency response swings that's your choice but don't mislead others down the wrong path.
And, more generally, the whole point I've been trying to make is what's required to publish data that can be reasonably compared between reviewers and devices. Any engineer worth the title will tell you the best way to do that is to control as many of the variables as reasonably possible to get consistent results. Published data that cannot be compared, or verified, is generally considered of little use. In the medical and scientific communities it's generally considered worthless.
If you test the 7 ohm output iPod with the Sony's IEMs above while the guy reading your results has the SuperFi's, what happens? He reads your review, sees the flat response, buys the iPod, and ends up with weird sound that's not even close to flat, and is disappointed. It's very misleading to do what you're suggesting--especially when both the impedance plot of the headphone used, and the output impedance of the device are complete unknowns. Anyone who understands math will tell you have too many unresolved variables to get anything resembling the final frequency response. If you can't see that I don't know what else to say?
[To convince me you would have to have some measurement that tells me something I didn't already know just from RMAA. Something surprising and useful, not merely something interesting in an academic sense.
Apparently hard facts, ohms law, etc. are not convincing enough for you? If frequency response swings of 8+ db are not "useful" enough, how about 13 dB more volume? Is that only academic to you as well? There are plenty of portable players, lots even, that can't drive plenty more low-sensitivity headphones to sufficient volumes. It's a very common complaint with some portable players. And 13 dB is huge.
But this has moved way off topic. I've been under the impression HydrogenAudio is about getting the facts right and being objective so that's all I've been trying to do. Someone else chime in if they want?
I'm not really sure how to interpret this response to my statement. Are you disagreeing with me about nonlinear mixing, or did you just quote the wrong block of text?
My initial answer concerned DAC non-linearities. I generalized it somewhat shortly after and now just realized that the new version has no relation to the quoted text anymore.
It also can't measure the famous pitch error of the Sansa.
Thats a fair point, although this is a pretty rare problem.
My question is whether or not one can detect the approximate 0.25% pitch error from one of the existing test results that RMAA already provides.
For example, the RMAA THD plot has a strong carrier @ 1.000 KHz, which if memory serves from the tests I ran, the Clip shifts to 1.0025 KHz. Using the expanded frequency scale feature that RMAA provids, I think that this error can be properly detected and estimated.
A similar argument can be applied to jitter. While 1 KHz is not an ideal frequency for the purpose (10 or 11 KHz would be better), audible amounts of jitter can be detected and estimated in a similar fashion from the existing 1 KHz tests. Furthermore, RMAA allows you to move the existing test signals aroound in the configuration menu.
Apparently hard facts, ohms law, etc. are not convincing enough for you? If frequency response swings of 8+ db are not "useful" enough, how about 13 dB more volume? Is that only academic to you as well? There are plenty of portable players, lots even, that can't drive plenty more low-sensitivity headphones to sufficient volumes. It's a very common complaint with some portable players. And 13 dB is huge.
You;'ve missed an improtant point. You don't need a $10K test set to show the effects of IEM load impedance on a digital player. All you need is RMAA, a $5 Radio Shack 3.5 mm splitter, and a pair of headphones.
If you want to test the effects of standard resistive loads, buy a pack of 15 ohm and/or 33 ohm resistors for aboout a $buck each, and solder them into some 3.5 mm male headphone plugs that you can get for I think less than $3 a pair at Radio Shack.
It also doesn't measure the DAC linearity (unless you count the -60 dB spectrum plot but virtually any DAC has near zero error there) or square wave response.
Sigma Delta DACs (the industry standard for about the past decade or more) can't possibly have linearity problems. That's one reason why virtually every DAC has low error at FS-60 dB .
I should probably give it up, but there are plenty of examples of Delta-Sigma DACs with linearity error.
I'm not at all convinced by unsubstantiated claims. I tested over 100 audio products with converters in them for my old PCAVTech web site (some were never posted), and I never saw a Sigma Delta converter that was appreciably more nonlinear than suggested by its noise floor. I tested some real garbage like the Avance semiconductor converters before Realtek bought them out and put them and us out of that misery.
Besides, there are virtually no products that drive their outputs with the true output of the DAC. They all add buffers or I/V converters in the same chip as the DAC or in another. There can be nonlinearity in them. But there is no doubt that RMAA tests for nonlinearity at both high and low levels.
When people talk about DAC nonlinearity, they are usually talking about lack of monotonicity and/or missing codes. Sigma Delta DACs can't have those because strictly speaking the don't decode the whole digital data word like a ladder DAC does.
I feel bad about beating up someone who is basically friendly to The Cause, but there is such a thing as going over the line and claiming omissions that aren't really omitted. At worst, RMAA has a credible FFT analyser that you can break out and use all by itself. Some of your plots by other means are just FFT plots that you are analyzing in your comments. You could do the same thing with RMAA.
It also can't measure the famous pitch error of the Sansa.
Thats a fair point, although this is a pretty rare problem.
My question is whether or not one can detect the approximate 0.25% pitch error from one of the existing test results that RMAA already provides.
For example, the RMAA THD plot has a strong carrier @ 1.000 KHz, which if memory serves from the tests I ran, the Clip shifts to 1.0025 KHz. Using the expanded frequency scale feature that RMAA provides, I think that this error can be properly detected and estimated.
At the time I wrote this I was having problems finding the results of my RMAA tests on my OEM-firmware Clip+. Well I finally found them and checked out the detailed FFT analysis for the THD test. The FFT plot very clearly shows that the 1 KHz tone is not centered on 1.000 KHz,. A quick visual estimate says that the 1 KHz tone was actually reproduced as 1.0025 KHz.
Therefore, the statement that RMAA "...can't measure the famous pitch error of the Sansa." is simply false.
For 99.95% of consumers this technology will be nothing more than a marketing gimmick.
And I really doubt it will take off in the nearest 10 years (or ever).
I feel bad about beating up someone who is basically friendly to The Cause, but there is such a thing as going over the line and claiming omissions that aren't really omitted.
We are on the same side of the cause and I'm sorry if I crossed some line. But I'm not sure what line that is? You put the above sentence in the DAC linearity reply. So I'll start there.
I'm not at all convinced by unsubstantiated claims. I tested over 100 audio products with converters in them for my old PCAVTech web site (some were never posted), and I never saw a Sigma Delta converter that was appreciably more nonlinear than suggested by its noise floor.
Fo DAC non linearity I was talking about the linearity of the entire device. Obviously, I'm not opening up the players and probing the raw DAC output which generally isn't available anyway with a "SOC" (System On a Chip) design. I have seen considerable errors, and I've also seen published Audio Precision reviews that document significant non-linearity errors. Here's an example of a -6 dB error at - 90 dBFS--the value I run my tests at:
(http://www.stereophile.com/images/archivesart/1109HRTfig4.jpg)
You can find the entire test here:
hrt-music-streamer-usb-da-converter-measurements (http://www.stereophile.com/content/hrt-music-streamer-usb-da-converter-measurements-0)
I don't know if that helps "substantiate my claim" for you, or not, but the above is the sort of error I was talking about. I consider a -6 dB error at -90 dBFS to be a significant, and measurable, linearity error that at least some would find interesting. Apparently John Atkinson at Stereophile agrees it's worthwhile to measure linearity as well because he does it regularly. If all Delta Sigma DAC devices, as you claim, have not had any linearity problems in the last 10 years, why bother to make the measurement? If you want to get into the semantics of terminology, and accuse me of incorrectly wording exactly what sort of error I'm talking about, fine. But I think we both know what sort of real world measurement I was talking about (i.e. the one above). What line did I cross here?
At the time I wrote this I was having problems finding the results of my RMAA tests on my OEM-firmware Clip+. Well I finally found them and checked out the detailed FFT analysis for the THD test. The FFT plot very clearly shows that the 1 KHz tone is not centered on 1.000 KHz,. A quick visual estimate says that the 1 KHz tone was actually reproduced as 1.0025 KHz.
Therefore, the statement that RMAA "...can't measure the famous pitch error of the Sansa." is simply false.
I guess the key phrase is "visual estimate". I haven't tried zooming in to the max amount on RMAA. But my earlier comment was that you won't see those results
directly in a table, etc. But I agree with your earlier statement that it's a rare error and not generally something of interest.
You;'ve missed an improtant point. You don't need a $10K test set to show the effects of IEM load impedance on a digital player. All you need is RMAA, a $5 Radio Shack 3.5 mm splitter, and a pair of headphones.
I didn't miss it at all. In the RMAA blog article I talk about the pros and cons of using headphones vs resistors as a load and recommend people use a load for testing with RMAA. I completely agree you can go buy resistors and I advise doing just that in my RMAA article. My debate with Saratoga was over his suggestions that (paraphrasing):
A - Output impedance doesn't matter to anyone so it's not worth measuring
B - All 16 ohm headphones are essentially similar enough to be used as a load for a product review that can be fairly compared to other product reviewsAs for "needing a $10K test set" I've never suggested anyone who wants to publish a review run out and buy an expensive audio analyzer. I have said many times here, and on my blog, RMAA is a great tool
when it's used properly. And I've tried to provide some useful advice to help people do just that.
My big issue with RMAA is it's often viewed as this sort of "Gold Standard" on many of the forums, etc. Many people just accept virtually anyone's published RMAA results as being correct. That's
far from being true. As I said earlier, you're not a typical RMAA user. You know how to use it correctly, most don't.
So the point of the "$10K test set" was to point out such analyzers are the "Gold Standard" not RMAA. If you really want to have complete, definitive, verifiable and repeatable measurements that conform established ndustry standards, RMAA is not the correct tool for the job. Yes, with enough knowledge, other hardware, other software, etc. you can kludge your way around some of it, but that's also not very practical for a lot of people.
I believe there are some significant things RMAA doesn't
directly measure or provide and/or you rarely (if ever) see in published RMAA measurements. They include:
- Test Levels For Repeatability and Comparability (a meter solves this, but again, it's hardly ever published with RMAA results)
- Maximum Output Power (requires a good meter and lots of trial and error with RMAA or other software and is rarely done)
- Output Impedance Of Headphone Outputs (can also be calculated if you have a known load, but few ever do)
- Linearity Error (requires other perhaps paid software and a quiet PC sound interface)
- Square Wave Response (requires other software)
- 11025 hz Standard Jitter Test (could be done with other software but I haven't seen any free analyzers with sufficient resolution)
My goal was to point out the common real-world problems with most published RMAA results. If you've used the software much, I'm sure you've seen it provide invalid results--there are several on my blog such as:
(http://3.bp.blogspot.com/-Fe0ZitkL33U/TVxNujcOqNI/AAAAAAAAABs/OY_2I28ssi8/s1600/hp%20310%20pc%203%20different%20loads%20results.png)
The above is the same PC headphone output, with 3 different loads, all calibrated correctly using RMAA's calibration dialog. RMAA did not detect any clipping or report any errors. Look at the THD and IMD for the SuperFi headphones.
RMAA reports 100+ times more distortion just from using different headphones. Why? It likely clipped due to the impedance swing of the headphones. But would the average person using RMAA know that? Here's the frequency response graph from RMAA for the same test run:
(http://1.bp.blogspot.com/-anEzUOt1CGs/TVxO5uzRi7I/AAAAAAAAAB0/7GV3WgVpauk/s1600/hp%20310%20pc%203%20different%20loads%20frequency%20response.png)
The SuperFi headphones (see the impedance plot I posted earlier in this thread) caused this wild frequency response which swings from +6 dB to nearly -9 dB. This is also another good argument for knowing the output impedance. I don't think anyone would argue the above frequency response change is inaudible or doesn't matter. RMAA got the graph reasonably correct but really messed up the distortion measurements. This kind of problem would be obvious on an analyzer, but it's rather hidden in RMAA.
And below is another actual run with RMAA. The 3rd column was the music player on my cell phone. It really doesn't have 237% IMD, honest. But RMAA thinks it does. There were no error messages from RMAA. It got the noise and dynamic range close to correct but fell on its face for all the other measurements:
(http://2.bp.blogspot.com/_gnU30vMbtYg/TUrgUXmHaWI/AAAAAAAAABM/h7TFcuvPn0s/s1600/rmaa-problems.jpg)
RMAA is basically a "magic black box" and you have almost no control over, or sometimes even knowledge of, what it does internally in the software. You just set the levels and click the GO button. It either works or it doesn't. And sometimes it doesn't, and sometimes it only half works. You can't change the test signals, or the measurement bandwidth of the distortion analysis, or anything else besides the calibration level. It's a rather limited tool in many ways.
For example, when I run RMAA with a PC interface that supports 24/192, does RMAA cut off the THD measurements at the audible limit of 20 Khz? Does it measure out to 96 Khz? Is it something in between? I certainly can't control it, and I don't even know what it is. Perhaps there are posts in a forum somewhere that answer these questions, but are they correct or someone's guess?
Are the noise measurements unweighted? Weighed? Bandwidth limited? Are they calculated out to the measurement limit? What if I want an A Weighted reading to compare to the dBA spec many manufactures use?
The program hasn't been updated in a few years, and the developer apparently has lost interest in it. It has some blatant bugs that many have complained about. I'm not trying to say it's useless, only point out its limitations and how it's very commonly used incorrectly or in ways that are misleading.
It's so popular because it's free and seems easy to use. But, in reality, it's also really easy to get bad numbers with RMAA. And to use it correctly, and overcome all the limitations of PC sound interfaces, not having absolute levels, gain matching, external divider networks, grounding, supplementing it with other software, etc., RMAA ends up being far more difficult (and error prone) than using a genuine analyzer.
And, back on topic, does anyone know if the newly announced iPad 2 has the much talked about 24 bit Wolfson DAC Apple is supposed to be switching to? I've tried my luck on Google and come up empty. Both are related to recent announcements, so you get plenty of hits, but nothing I've found so far mentions the iPad 2 itself having 24 bit support.
The sole audio performance spec on the Apple iPad 2 Tech Specs page is a useless: "Frequency response: 20Hz to 20,000Hz". And under supported formats, there's no mention of 24 bit:
Apple iPad 2 Specs (http://www.apple.com/ipad/specs/)
I feel bad about beating up someone who is basically friendly to The Cause, but there is such a thing as going over the line and claiming omissions that aren't really omitted.
We are on the same side of the cause and I'm sorry if I crossed some line. But I'm not sure what line that is? You put the above sentence in the DAC linearity reply. So I'll start there.
I'm not at all convinced by unsubstantiated claims. I tested over 100 audio products with converters in them for my old PCAVTech web site (some were never posted), and I never saw a Sigma Delta converter that was appreciably more nonlinear than suggested by its noise floor.
Fo DAC non linearity I was talking about the linearity of the entire device. Obviously, I'm not opening up the players and probing the raw DAC output which generally isn't available anyway with a "SOC" (System On a Chip) design. I have seen considerable errors, and I've also seen published Audio Precision reviews that document significant non-linearity errors. Here's an example of a -6 dB error at - 90 dBFS--the value I run my tests at:
(http://www.stereophile.com/images/archivesart/1109HRTfig4.jpg)
You can find the entire test here:
hrt-music-streamer-usb-da-converter-measurements (http://www.stereophile.com/content/hrt-music-streamer-usb-da-converter-measurements-0)
I don't know if that helps "substantiate my claim" for you, or not, but the above is the sort of error I was talking about. I consider a -6 dB error at -90 dBFS to be a significant, and measurable, linearity error that at least some would find interesting. Apparently John Atkinson at Stereophile agrees it's worthwhile to measure linearity as well because he does it regularly. If all Delta Sigma DAC devices, as you claim, have not had any linearity problems in the last 10 years, why bother to make the measurement? If you want to get into the semantics of terminology, and accuse me of incorrectly wording exactly what sort of error I'm talking about, fine. But I think we both know what sort of real world measurement I was talking about (i.e. the one above). What line did I cross here?
I don't think that the above shows "DAC error" as such.
First a little history, which you may or may not be aware of.
When digital audio first came generally available the complaint "measures good but sounds bad" was often heard. In retrospect, it is not clear that all of the complaints were based on reliable listening.
At any rate, there was supposedly bad-sounding digital equipment that had great frequency response, low noise and low nonlinear distortion that "didn't sound right". One actual technical difficulty was that the resistor ladder DACs of the day were prone to errors of a kind that had no direct equivalent in the analog domain. Therefore, means were devised to test for them. One of these "new forms of distortion" were due to errors in the resistor ladders that lead to incorrect step sizes when an ascending or descending sequence of digital values were applied to the DAC. Isolated steps could be so wrong that the desired staircase was had steps that either didn't exist or even went in the wrong direction. This is known as lack of monotonicity.
While Stereophile has tested for this kind of error for decades, AFAIK no actual instances equipment with this fault it have been published.
My review of the Stereophile archives suggests to me that by the time they had the resources to adequately test digital gear, the new digital gear had improved to the point where monotonicity failure was rare or non-existent in high end audio gear. DACs based on resistor ladders were passing from the marketplace, and/or such resistor ladder based equipment as was being built in the high end price category was too highly perfected for gross errors of this kind to appear.
The review you cited does not appear to show difficulty with classic "DAC Error". Instead, it shows a product with relatively high nonlinear distortion of a fairly ordinary kind. If there were actual monotonicity errors, then there would be large steps and discontinuities in Figure 4 of the review you cited. Instead I see the effects of a gently curved transfer function which is confirmed by figures 6 and 7.
If there was classic "DAC error", the nonlinearities shown in figure 4 would have some dramatic breaks or steps. Instead it is just curved. It would not be as nonlinear in terms of traditional harmonic generation and IM as is shown in figures 6 and 7. The source of the nonlinearity could be the I/V converter or any buffer stage that followed it.
I believe that the far less costly Behringer UCA 202 would generally outperform the HRT Streamer. I suspect that one possible purpose of the Stereophile review was to show the inadvisability of buying low cost equipment. It might be might that they could have gotten something better if they spent far less! ;-)
There are renewed rumours that Apple will start selling 24-bit, 96 kHz or 192 kHz audio. Robert Hutton claims (http://robertmusic.blogspot.no/2014/03/led-zeppelin-and-future-of-hi-res-audio.html) it will launch at the beginning of June with the release of three Led Zeppelin remasters. Pricing is supposed to be one dollar above the current iTunes Plus.
Not sure where he got his info from. He somewhat supports his claim by noting that certain labels have stopped catering to HDtracks. He also argues that Apple has already got the largest HD catalog of HD audio, as they've requested labels to upload HD content for a number of years.
EDIT: German tech site heise.de sees (http://www.heise.de/newsticker/meldung/iTunes-soll-angeblich-HD-Musik-bekommen-2168193.html) other signs of a major revamp of the iTunes music store which could include HD audio:
1.) Apple is increasingly concerned about losing sales to Pandora, Spotify, and friends. And it doesn't look as if iTunes Radio is gaining any traction. (At the very least, it doesn't generate significant *sales* for the music store.)
2.) Neil Young's crowdfunding campain was a success. So maybe the momentum for HD audio is there.
3.) A launch date in early June would fit well in Apple's calendar. WWDC runs June 2 through 6.
4.) Apple is hiring in several areas for the iTunes music store.
I'm not sure if that's accurate or not. At this point, anything not officially announced by Apple has to be viewed as rumor or conjecture. It would be nice if Apple started selling lossless content (I could care less about placebophile 24-bit, 96KHz/192KHz content) as I could then stop buying CDs all together. However, if the rumor is true, I'm still confused on their pricing. Is it $1.00 extra per track or $1.00 extra per album? Hutton's blog makes it appear that it's $1 per track meaning a 12 track album won't cost $9.99 but rather $21.99. Other albums, like Led Zeppelin IV, would cost $17.99 and albums like 1 from The Beatles would cost $39.99. Those prices seem a bit excessive even for the iTunes Store which normally sells albums at the going rate ($9.99) or $1-$3 more compared to Google Play and the Amazon mp3 Store. If Apple did launch such a service and the price rumor was accurate, they would be pricing themselves out of the market. An extra $1 per album would make more sense.
Then again, this has to be taken with a big grain of salt seeing as how the whole "lossless on the iTunes Store" rumor has been circulating for the past 4 years (if not more).