Apple TV versus Squeezebox/Audio Streaming

Discussion in 'Apple TV and Home Theater' started by ziggy1968, Mar 17, 2008.

  1. ziggy1968 macrumors newbie

    Mar 17, 2008
    Hi there.

    My first post, so I hope I'm not doing anything wrong! Here's my problem I need help with....

    I want to stream Audio from my iMac to the Apple TV across my hard wired home Cat5 network.
    All my music (7000 songs) is imported in to iTunes in uncompressed AIFF format. I do this because I have a very serious Hi Fi.
    I presently use a device called a "Squeezebox" The connection is Analogue stereo Phono out, then in to the pre-amp via L&R phono.
    I want to replace the Squeezebox with the Apple TV.
    I plan to connect the Apple TV via HDMI to my Sony KDL-40X2000 TV,
    via Opitical( TOS link) output to my Bose home cinema set up, which is a totally separate system to my Hi Fi (horses for courses etc)
    and via the RCA (Phono) outputs to my Hi Fi Pre-Amp.

    I need to know that the audio that I stream into the Apple TV ie AIFF, CD quality audio(Sample size 16Bit, Sample Rate 44.100 kHz & Bit Rate 1411kbps) is what will come out of the analogue RCA outputs. I have no reason to believe it won't be, I'm sure Apple will use decent Digital Audio Converters, but there is no way of knowing for sure, as there is no information on the Apple web site.

    By the way, I don't believe Apple Lossless is "CD" quality, so I have no interest in Airport Express.

    Finally, Can anyone tell me what the power consumption is of the box when it's in use and also when it's in standby. Again, this information is unavailable. Strange as it may seem, there is no on/off switch and I need to know if I should be pulling the plug out at night, to not waste energy.

    Many thanks in advance
  2. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    Even if I owned one I would know how to figure this out. Aside from testing it with a multimeter I guess. I have a squeezebox, I find it easier and quicker to use than my Mac Mini Media computer (very similar setup to AppleTV).
  3. Alrescha macrumors 68020

    Jan 1, 2008
    Ummn.. I'm curious - what part of 'lossless' don't you believe? You think that the output of an Apple Lossless file isn't bit-for-bit identical to the original? Why not? That's the whole point of 'lossless', after all.

  4. theBB macrumors 68020


    Jan 3, 2006
    Delusion? Conspiracy Theories? :)
  5. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    No No No, its much more simple than that. Audiophilism.
  6. theBB macrumors 68020


    Jan 3, 2006
    By the way, I don't think AIFF files can hold tags or artwork. Besides taking up half the space (and being lossless), that is another advantage of Apple Lossless.
  7. Avatar74 macrumors 65816


    Feb 5, 2007
    How serious? $1000? $2000? $200,000? But I'll get back to that in a minute...

    A cursory examination of the throughput over an AirPort Extreme base station using SNMP, and an examination of the signal originating from the TOSLINK output on the AppleTV confirms that:

    a) The signal being transmitted is sent without additional compression over the network. That is, 256 Kbps AAC is sent at 256 Kbps, 1411 Kbps Linear PCM is sent at 1411 Kbps, and so on.

    b) The AppleTV reconstructs any source as 16-bit Linear PCM upon transmission to a receiver. That is, no matter the source, it is reconstructed as CD Digital Audio (Red Book specification) before transmission out of the AppleTV to your playback system.

    That being said, 16-bit stereo Linear PCM is not fundamentally discernible from 128 Kbps AAC. I don't care what you've read in umpteen audiophile forums and magazines. They don't know what the hell they're talking about. The Audio Engineering Society is the only organization I am aware of that conducted listening tests meeting SCIENTIFIC standards, and those tests confirmed that 128 Kbps AAC is acoustically transparent relative to 16-bit Linear PCM.

    If your library consisted primarily of 24-bit Linear PCM, I would argue in favor of maintaining the uncompressed original data. The reason for this is simple... the amplitude resolution of 24-bit Linear PCM vastly exceeds that of 16-bit (2.7 million amplitude values per quantization interval with 24-bit versus 65,536 amplitude values per quantization sample with 16-bit). To put it another way, nevermind 128 Kbps, the gross inferiority of CD audio becomes wholly apparent when compared to 24-bit Linear PCM. Mind you, I'm not making an argument in favor of analogue either... pumping amplitude to compensate for a high noise floor and thus creating a false perception of a "fatter, warmer" sound introduces more problems than it solves.

    Furthermore, 16-bit Linear PCM is so prone to quantization interval errors that dithering is required in the mastering process which ultimately increases the noise floor. Even though this increase is arguably imperceptible, it amuses me that audiophiles do not summarily balk at all CD audio as being utterly horrid relative to the pristine quality of 24-bit Linear PCM audio. Instead, they spend thousands and thousands of dollars trying to purchase all kinds of overinflated transports and hardware to milk a level of fidelity out of CD's that just isn't there.

    That being said, it is not difficult at all to reconstruct the frequency response and amplitude resolution of 16-bit CD audio with 128 Kbps AAC, given that AAC, developed jointly by Fraunhofer-IIS, Dolby Laboratories and the Motion Picture Experts Group, is a perceptual coding schema that uses methods that extend beyond that of Adaptive Delta PCM in terms of the ability to eliminate data requirements in the process of reproducing the same analogue wave. That is something audiophiles do not seem to grasp.

    Airport Express aside, you're wrong. Apple Lossless, like other lossless formats (ADPCM being one), uses linear prediction as a method for reducing the data required to reconstruct the exact same analogue wave. You do understand, don't you, that you cannot hear the digital data, and it is merely used to reconstruct an analogue signal... and that even 16-bit PCM is missing a tremendous amount of data but manages to sufficiently reconstruct an analogue wave within the limits of human perception?

    I'll give you an example...

    Adaptive Delta PCM differs from standard PCM in this manner: PCM normally records the amplitude value at any given sample interval as a constant 8-, 16-, 24-bit and so on absolute value. So if, let's say, the value at one interval is -28 dbFS, and then the next sample is -27.59dBFS, both of these double-integer values will be stored in their complete form... and stored in the same wordlength, wasting a certain amount of data that isn't required. But with the quantization throttling of ADPCM, only the relative CHANGE in value is stored... In other words, the value of 0.41 (the difference between those two amplitude values), is stored, requiring far less data.

    Now do you understand that from this, across a series of discrete time sampled audio... the EXACT SAME analogue wave can be reconstructed?

    There are other issues such as 20kHz lowpass filtering used to eliminate aliased frequencies and so on that I won't go into but I would strongly recommend a cursory read of Ken Pohlmann's Principles of Digital Audio. Its first edition in the early 1980's, Pohlmann's book is an engineering bible and a seminal read for anyone who wants to really understand how digital audio works... Take all your audiophile forums, and articles and issues of Stereophile and please, PLEASE throw them in the trash. The words of these disinformation jockeys are not worth the price of the paper they're printed on.

    Simply put, you will not be able to tell the difference between 128 Kbps AAC and CD audio... I've gone up and down this debate and not one audiophile can produce a single scientific study published in a peer-reviewed scientific/engineering journal to prove otherwise. It doesn't matter how "serious" your sound system is. Your ears cannot discern a difference because the actual difference between the analogue wave reconstructed from 16-bit PCM and 128 Kbps AAC is below the threshold of human perception.
  8. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    I was waiting for a response like that.
    BTW Avatar how do you know all of this stuff? Not refuting, just curious/interested. From my own education in PsychoAcoustics I have long been aware that audiophiles (whatever that term means) are only capable to a certain limitation (i.e. their sound reproduction hardware and their own ears) to decipher differences between encoded audio. I understand that at a certain point you are just not able to increase fidelity of audio based on the source (audio CD) not having enough information. Live music remains the only way to achieve a perfect listening experience, but then the room effect of course can then screw that up. But that's a different story.
  9. Avatar74 macrumors 65816


    Feb 5, 2007
    I've worked a lot with audio over the years and did some professional CD mastering. Also, one of my close friends was an engineer at CBS Soundlabs... he did some research contributing to their FMX patent which was an attempt at developing a FM quadrature. He told me a hilarious story about how he was given a tour of the Wadia factory (I know a guy who bought one of their $3500 CD players)... on the manufacturing line he came to find with his own eyes that the transport mechanism (the laser pickup and motor assembly) was supplied by Pioneer, who puts the exact same units in their (then) $150 CD players.

    You could also say that my parents being scientists had a lot to do with my appreciation for factual observation rather than hearsay, anecdote and (the worst form of evidence) testimony... That same background would certainly explain my love of the show "House." :D

    The thing I find wrong with most ABX tests is that they generally seek to get the listener's opinion on which format sounds "best"... but that introduces all kinds of potential for bias for several reasons. The sample population is never random because it's usually audiophiles seeking affirmation who find these forums and ABX tests online, not researchers going out to randomly select people without any knowledge of their background in order to produce a true microcosm of the general population.

    Another problem is that these ABX tests are not designed to actually verify the listener's ability to discern one format from another. Their design inherently tests which format the user thinks sounds best. Consider carefully how different that question really is... It's like the difference between the easier contextually-biased multiple choice questions and the much harder, context-absent fill-in-the-blanks questions on an exam. Even so, in comparisons of repeated trials from different sources there are no statistically significant findings that show a consistent preference for one format above the rest. And, as is the case with the Hydrogen Audio formats, they're comparing one lossy compression schema to another in totally uncontrolled settings.

    What would be appropriate is a controlled setting with one set of speakers (not umpteen different setups between every listener on the web), a randomized sample population, testing administrators who arent aware of which number sample is which format (the double-blind aspect), and, most importantly, a requirement that the user be able to actually identify not which sample sounds better, but which sample is in what format. If they can consistently identify the PCM sample against the AAC sample greater than 80% of the time in repeated trials, with one listen of any one song (as opposed to comparing the same part of the same song over and over; remember, we are not looking for whether they can tell the difference after listening to the same 10 second clip a thousand times... that isn't a typical listening circumstance), and see these results replicated by different populations in different trials in different clinical settings, THEN we've got something statistically significant to discuss.

    Otherwise, the claim is bunk.
  10. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    Nice, I work as an Acoustical Consultant (MS in Acoustics) and have had many a discussion regarding the resolution of our own ears to be able to detect any differences in 384bitrate and 256bitrate only to be written off that my hearing is bad and that "others" are capable of hearing a difference. I agree that published audiophile literature is a bunch of snake oil.

    Needless to say I still think a SqueezBox is easier to use than an AppleTV.
  11. Avatar74 macrumors 65816


    Feb 5, 2007
    Depends on what you consider "easy"... Two big advantages of AppleTV is a large onscreen GUI for readable navigation compared to a small remote control screen, and the ability in version 2.0 to use AppleTV as an AirTunes port which gives you the ability to do more complex searching, playlist generation and queueing from any computer on the network. I am also relatively certain that Apple is exploring the possibility of using iPhone/iPod Touch as a navigation device for AppleTV... a multitouch remote which is infinitely more usable than a scrollwheel-guided navigation system.

    But, this little gem had me rolling, regarding the $2000 Squeezebox Transporter "audiophile" music player:

    First... I can't think of a digital transmission system since maybe the mid-1980's that isn't "bit perfect" because of error correction inherent in optical-digital transport.

    Second... the word "jitter" makes me gag. Jitter, contrary to popular audiophile misconception, has not been a problem since the mid- to late-1980's with the advent of internal reclocking of the signal, improved sample & hold buffering at the DAC and the now commonplace nature of extraordinarily accurate quartz oscillators. Ok, sure, they might lose more time than a cesium fountain atomic clock but for the purposes of 48kHz sample accuracy, let's just say any artifaction is going to be below the threshold of human perception.

    "Clock signals in Transporter are handled not as ones and zeroes, but as precision analog signals." ... I think they mean "voltage oscillation" but if they said that, the utter absurdity would be revealed. EVERY quartz oscillator creates an electrical oscillation i.e. "analogue signal". Almost every piece of audio hardware and computer hardware has a quartz osscillator keeping time... and each is pretty much as accurate as the next. This site is so chock-full of bullcrap I'd strongly encourage anyone with two brain cells rubbing together to stay far, far away from it.

    Utterly hilarious read, though, that Squeezebox site. Aside from the fact that Logitech (who makes the lowest quality speakers, joysticks and so on that I've ever seen) decided to enter into the snake oil business, they use the word "audiophile" in their marketing so liberally it should be replaced by the word "sucker".
  12. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    When comparing two interfaces, I consider the easier of the two to be the interface that is less difficult to get from startup to playing. I know this sounds very ticky-tack, but the Squeezebox doesn't require a TV at all, and when you're watching TV, or have a BBQ going on, or want anyone else in the room (that has no idea how to switch your video port on your TV) to change the song. The SqueezeBox is fast and its large screen (I have an older version) makes it very user friendly. Though I'm surprised at the price of these things :eek:

    As for the "SB Transporter" thing, I guess you should get that if you want "Astounding Analogue"..... whatever the hell that means.
  13. ziggy1968 thread starter macrumors newbie

    Mar 17, 2008
    Wow! Thanks for all the information guys

    I really didn't mean to start a war out there!

    a) The signal being transmitted is sent without additional compression over the network. That is, 256 Kbps AAC is sent at 256 Kbps, 1411 Kbps Linear PCM is sent at 1411 Kbps, and so on.

    b) The AppleTV reconstructs any source as 16-bit Linear PCM upon transmission to a receiver. That is, no matter the source, it is reconstructed as CD Digital Audio (Red Book specification) before transmission out of the AppleTV to your playback system.

    That's the info I wanted and I still don't understand why Apple couldn't/wouldn't provide it, so many thanks for your help there.

    My background is as a live sound engineer. I first came to America in the early 70's with a band called Gentle Giant (Yeah, I really am that old!) and returned many times with bands you may know a bit better. The Kinks?

    When I decided to put all my vinyl in to the loft (or attic ) and re buy my music collection on CD, it was purely for convenience. I still prefer the sound of vinyl ( I think the best explanation was by a Japanese guy I know who uses old valve amps to power his speakers: "It makes me feel warmer") illogical, but I kind of get it.

    My Hi Fi has a very "Live" sound and consists of large floor standing Bi-Amped speakers powered by professional amps. It runs at around 750 Watts Rms a side. It's LOUD.

    Once you go down the digital path there is no turning back, so When I Tunes came along and streamers became more common, I investigated. I opted for the Squeezebox route to replace my CD "Jukebox". I got a bunch of my friends round (all professional audio guys in different fields, Studio, Live, Installations etc) and over a few beers we did numerous A/B comparisons.

    None of us could hear any difference between AIFF via the Squeezebox and the direct source from the CD player. However, we could all hear a difference with the MP3 files (much to my sons annoyance, who did the double blind tests!) we didn't test the Apple lossless as it wasn't out then.

    Any kind of signal processing has to do something to the sound by definition, surely? So while you are probably correct in saying it is not lossless (technically speaking) why bother processing the signal ie compressing it and then uncompressing it, when there is no need?

    After all, storage is cheap and I've got 7,000 songs on my 400 Gb Glyph (top quality hard drives by the way) and when I get my new iMac I can get them all on the hard drive (1Tb) with ease.

    The one thing I learned when doing live sound is that you strive to do as little as possible to the original source. It pains me to see these young guys who come out of college courses, piling on EQ to a snare drum, instead of going to the stage and trying to re tune the drum itself.

    I've since set up the Apple TV and it works just as well as the Squeezebox sound wise, the big difference though is that I can see what I'm doing. The other gentleman who said the Squeezebox screen is large, must be younger than me and with better eyes! I can see all the artwork as well.

    In the end I've never met anyone who can explain technically why JBL speakers sound better on Rock music and Tannoy sound better on classical. They do though!

    Many thanks for your help gents, it's most appreciated.
  14. GimmeSlack12 macrumors 603


    Apr 29, 2005
    San Francisco
    The Kinks, never heard of them ;)

    Either way, on any sort of web forums (this one in particular) we get a lot of folks that talk big but really have no idea what they're talking about. So naturally without verification, we attack them :) (seems fair right?). I'll read over your post and see if I have any additional tips for you.
  15. Alrescha macrumors 68020

    Jan 1, 2008
    Compressing something digitally is not the same as compressing something in the analog domain. When you use Zip compression on a file on your computer and later unZip the file, you get exactly the same file that you started with. I'm no expert on Apple Lossless, but it is my understanding that what you get out is what you put in.

    If disk space is not a concern, then by all means don't bother to compress. :)

  16. Avatar74 macrumors 65816


    Feb 5, 2007
    Oh hell yes The Kinks. I'm in 30's but my older cousin used to play records to get me to sleep while babysitting when I was 2-3 years old. Can I just say that sound engineers today do not possess half the talent of the old-timers. There's no sweetening... it's all amplitude pumping to constant clipping, distortion, garbage. Very few albums these days take advantage of the dynamic range of the digital medium.

    I've done both live engineering and studio... studio is a whole different deal. Both are crafts that take skill, but there are different objectives and different considerations. e.g. With PA you have one chance to get it right. With studio recordings, problems with fidelity are going to be much more noticeable and permanent once the master recording is final.

    I certainly, and greatly, respect your work as a live sound engineer... particularly one in an era with a lot more attention to craftsmanship than I see these days, but I'll tell you that digital formats in the mastering world are a very different ball of wax. That being said, let me try to address some of your questions...

    Again, this goes back to a few tradeoffs. Vinyl doesn't possess the dynamic range of CD audio. It's maybe about 16 decibels short. Ironically, that's bad news for jazz and classical music... the two genres that audiophiles LOVE to listen to on vinyl.

    In the early days of digital mastering, more than 20 years ago, many recordings were lacking on the low end of the frequency spectrum. There are a number of reasons for this, but since then both the hardware, software, and engineering techniques have adjusted to compensate for the fact the noise floor itself is virtually absent. The noise floor of the analogue medium does contribute to the warm, fat sound... consider that you've got a constant underbelly of low frequency "fuzz" plaguing the entire recording.

    Therefore what you're actually hearing an analogue recording is false... even though it may "sound better". But I'll tell you something, the first time I fired up my optically-linked surround system and played "The Thomas Crown Affair" and heard in the opening dialogue an absolute dead silence between lines, I was elated. The noise floor is so imperceptible that it adds to the ability of any sound system to make a more faithful reproduction that sounds, if properly engineered, like it's right there in the room.

    I'm a big fan of LOUD for PA... I used a tri-amped (1600 watts total) Peavey Project II system to disc jockey a school dance once. It was the first time I'd ever heard middle school kids complain about music being too loud.

    But where sound reproduction from a recorded medium is concerned, I'm a bigger stickler for accuracy of reproduction. I'm glad you mentioned Tannoy. They make GREAT reference monitors. I'm assuming your British since that's where Gentle Giant came from, and the Kinks... Well, the Brits know how to make great speakers. I have some KEF speakers and love the hell out of them.

    Well, there's the rub... MP3 is a VERY different codec from AAC (MPEG-4, Part 10 spec). MP3 performs very poorly at low bitrates due to its dependency on older compression techniques and lack of more sophisticated perceptual encoding. AAC, on the other hand, is indistinguishable from CD audio at 128 Kbps. You are absolutely right you can hear differences between MP3 and AIFF... but even without a few beers in your system you won't tell apart 128 Kbps AAC and 16-bit LPCM even with the goldest of golden ears.

    Welllll... Let me straighten a few things out. Signal processing is one thing. Digital compression is another. And perceptual encoding is a little bit of both and also neither. As another user put it, compression in the analogue realm means something considerably different from data compression in the digital domain. We're not talking about compressing the amplitude modulation of an analogue waveform. Instead, complex algorithms are used in the decoding hardware and/or software to recognize clusters of data such that the data could be truncated with a shorter data. Information can also be stored in a nonlinear fashion... entirely unlike the analogue realm.

    Imagine that you have a sequence of 12301234. Now imagine that you can use a value let's call x to represent the cluster "123". What a codec can do is reduce that string from 12301234 to x0x4... where x is understood by the codec to mean "123". It could go even further to say, in some manner, to truncate x0x4 to essentially say in an even smaller string that: x appears twice, and is interrupted by a zero and a 4.

    Another technique, linear prediction, can reduce data in a manner similar to this...

    Imgn w rmvd ll th vwls frm th sntnc.

    The sentence can still be reconstructed if the algorithm of the decoder knows everything it needs to know about speech synthesis to fill in the blanks. Consider for a moment that your brain already did it... I doubt it was very difficult at all for you to decipher what the above sentence says.

    I know it sounds counterintuitive because this is a lot for a system to have to know... but we're talking about reducing data requirements of the medium, not the playback system. Notice the difference here... You're packaging more knowledge into the hardware so that the format can be made more compact.

    Then there's the aforementioned ADPCM (from my previous post) which uses techniques like quantization throttling and relative amplitude values to store only the change between amplitude values from one quantization sample to another, and limits the size of each sample to only the number of bits required to store the value given at that sample.

    All of these methods, in principle, can be used to reconstruct an analogue waveform with enough resolution that your ears cannot tell the difference between the reconstructed wave and the original. Now keep this in mind:

    in the 12301234 example, x0x4 is not what is played back. x0x4 is decoded, and, here's the important part, reconstructed into 12301234.

    Same with the sentence... the sentence is not played back with vowels absent, the algorithm is applied, the string decoded, and the original analogue sentence reconstructed.

    It is upon reconstruction where the potential for error can occur... but this potential is mitigated heavily by today's methods of digital sampling and reconstruction, encoding and decoding.

    One example is frequency aliasing. If you use 44,100Hz as the digital sampling frequency (again this is not an analogue wave but a representation of the number of times, frequency, a discrete time sample is recorded as digital data... a sample/quantization interval)... then according to Harry Nyquist's ever reliable equation, your frequency response extends to 22,050Hz. Why? Because the bare minimum data needed to reconstruct a sinewave is the peak and the trough of one wavelength, i.e. two digital samples per cycle (1 Hz). 44100Hz was picked because it gives enough overhead past the A-weighted spectrum... the range of human hearing.

    But what happens if you sample a frequency HIGHER than 22050Hz... you will get aliasing. Now, some people hear the term "aliasing" and think this means a jagged, staircase like signal... and audiophiles think that this is actually perceptible. Well, yes and no.

    First, aliasing is better understood by its dictionary definition which is closer to the truth of what it means in the digital realm... Take a look at this demonstration to see how sampling a 7000 Hz frequency 8000 times a second will actually create an aliased frequency of 1000 Hz in the recording.

    So what happens if we sample a 33kHz signal at 44.1kHz, we'll get an alias frequency because we aren't sampling the amplitude values of a 33kHz signal at the peak and trough... but somewhere else. The result is a frequency reproduced closer to 11kHz.

    What's the answer? An antialias filter. In the mastering stage, a 20kHz lowpass filter will knock out any frequencies that would not be sampled properly at the sampling rate of 44.1kHz. Problem solved.

    Just as there are scores of techniques used in analogue sampling, recording, mixing and mastering to maximize the potential accuracy of that reproduction... same goes with digital.

    One of the beauties of digital nonlinear editing is that you can do all kinds of nondestructive filtering, eq-ing, leveling, effects, etc. that do absolutely nothing to the source data... you can always go back to the pristine, unaltered source. But I know what you mean. At the same time, where do we draw the line between one kind of art and another? When it comes right down to it, the acoustic drum and trumpet are still artificially constructed instruments designed to reproduce sounds we first heard in nature. Even though the designs have changed, the instruments are still man-made as they always were.

    Andy Johns and Peter Collins have both used miking techniques to enhance the liveness of the drums on Led Zeppelin, Van Halen and Rush tracks. Kashmir by Led Zeppelin would not be the song that it is without the stereo phaser effect that Bonham threw on the drums... that one effect changed the vibe of the whole mix. When do we realize that a drum machine is simply a different kind of instrument than a live drum kit (I'm partial to real percussion but only because I practiced more on a real kit and that's what my mind is partial to).

    The problem isn't that the instruments are changing. The problem is that some people are just lazy. What you have to do is get out ahead of the technology and decide that you're going to master the new instruments in a way that no one else has... Talvin Singh is a hell of a drum programmer but he is also a phenomenally accomplished tabla player.

    The same goes for the recording medium. Digital has changed the game and we can sit and whine about it or we can figure out how to get out in front of it and take advantage of everything it has to offer in the way of reproductive accuracy.

    Agreed. This is my gripe about the Squeezebox... It's designed well for someone who just wants a listening room with no television but most people these days have an integrated home theater of some sort. It's a lot easier to navigate from 10 feet away with the AppleTV than even looking at a tiny screen in my hand.

    Amar Bose probably could. He contributed greatly, despite what the snobs want to say, to the advancement of sound reproduction. Though his goals are more oriented toward efficient output rather than accurate reproduction, this psychoacoustics professor at MIT from my native India knows a tremendous amount about how the brain perceives sound and could surely identify why JBL is better oriented to rock and Tannoy to classical. Although my immediate reaction is that rock tends to be heavy in bass and treble, and JBL makes simple speakers that produce tremendous SPL output in two ranges of the spectrum, whereas Tannoy follows my philosophy of smaller and more drivers to tend to the full range.

    If you've ever sat in a 2001-2005 Mercedes S-class and listened to the 13-speaker Bose Beta 2 system you'll know what I mean. The system uses smaller drivers, but a large number of them, along with custom developed acoustic waveguides modeled specifically to the Benz cabin's acoustics, to saturate the cabin with sound without exploding your eardrums.

    One thing JBL doesn't do very well is produce full-range response at very low sound pressure levels... They're mostly a PA speaker. Tannoy, KEF, Paradigm, B&W, etc. do a phenomenal job of maintaining frequency response at low sound pressure levels. This is critical in both instrumental recordings and feature film soundtracks, which make fuller use of dynamic range than the typically amplitude-pumped, distorted to hell rock recordings that tend to have two volumes... off and eleven.

    This is why I'm a big advocate of 24-bit Linear PCM audio for critical listening... the dynamic range of 16-bit CD's is only about 96.7dB whereas 24-bit LPCM extends up to around 140dB dynamic range with an amplitude resolution of (sorry I understated this in my previous post) 16.7 MILLION possible amplitude values per sample interval. I've done some instrumental recordings where CD audio just KILLS your ability to hear simultaneously quiet and loud sounds or abrupt shifts. Note that amplitude resolution, and not frequency response, is the real reason behind the inability to record cymbals very effectively. Audiophiles love to use cymbal heavy recordings as an example of the inferiority of certain digital formats. But they fail to understand that sampling frequency is really not the issue... Cymbals hover somewhere between 7 and 10kHz, both well within the limit of a 44kHz format such as AAC or 16-bit CD audio. But look at the analogue wave of a cymbal... the frequency is pretty constant, what's erratic is the amplitude! So erratic that 16-bit resolution just doesn't cut it.

    For critical listening I would like to see formats like DVD-Audio gain popularity... I know they'll always reside in the minority, but bandwidth and storage in nonlinear computer-based systems is getting cheaper and cheaper, and 24-bit LPCM shouldn't be hard to support going forward.

    That being said... For most listening, AAC does just fine... because the source recordings were all mastered to 16-bit CD audio anyway. Nothing more than 128 Kbps AAC, maybe 192 at best... is really needed for the vast majority of sound recordings out there.
  17. bgalizio macrumors member

    Apr 28, 2006
    I don't claim to be an audiophile by any means, just a music enthusiast. However, I disagree with your claim that 128 Kbps AAC is indistinguishable from 16/44.1 PCM. I have participated in quite a few blind tests and have 100% of the time (along with many others who played the game) picked the PCM "out of the crowd" and identified it as sounding better to me. The kicker? I have a high frequency hearing loss and listen on a modest system (maybe $2000 total).

    I agree that some people can't tell the difference, but many of us can. I think it depends on quite a few factors, though. Here are some, not in any particular order:

    1. The system
    2. The type of music
    3. The quality of the originally recorded music - ie: something "radio friendly" that is compressed to all hell and recorded craptastically may show little to no difference, but a great classical or jazz recording may show stark differences.

    Back to the original topic - the Squeezebox is my source of choice due to the ability to play FLAC audio at 24bit. Not sure if it's the DAC, but it also sounds better to me than my CD player.

    Edit to add: I don't own an Apple TV, so cannot comment on the sound differences between it and a Squeezebox. I am thinking of an Apple TV one day for streaming home video and photos, but probably won't use it for audio (though I will test the output against the Squeezebox, just for fun).
  18. Avatar74 macrumors 65816


    Feb 5, 2007
    Ok, where are the results published? If the answer is not "in a peer-reviewed science/engineering journal" then scientifically speaking your claim has no merit. Telling me you can tell the difference is not the same as having the veracity of the results withstand the scientific scrutiny of the peer-review process which would unequivocally substantiate your claim. I'm not saying anecdotes and testimony are worthless... but they have no merit in a scientific court of inquiry. Ten thousand people can claim to have seen the virgin mary on the side of a building and yet they can all still be in error as to the actual nature of the phenomena.

    If you did participate in an actual scientific study conducted by scientists/engineers at an academic or clinical institution, please provide their contact information so I can assess the veracity of the claim.

    Your anecdotes or testimony do not count as evidence that you can in fact identify a PCM stream 100% of the time in a randomized, controlled, double-blinded setting.

    Again, anecdotes and testimony do not count as evidence in a scientific court of inquiry. Produce the published results... I'd like to see where the test was conducted, what were the factors, who conducted it, what the scientific controls were, how many science journals they submitted their findings to for publication, and how many science journals accepted and published said results.

    It should not depend on such factors. The sensitivity of human hearing considerably surpasses the variance in systems, music and sound reproduction quality that if you cannot perceptibly hear the difference under less than absolutely pristine and ideal conditions (do we also need to play it back in an anechoic chamber?) then the difference is even more likely to be statistically insignificant. That being said, no one has ever published results in any science or engineering journal that would substantiate an ability to tell the difference on the most pristine of systems with the perfect master recording in ideal acoustic conditions.

    That being said, the data and methods used in AAC and PCM are both above a threshold of distinguishability.

    Consider that your eyes can detect a shift in color wavelength of one nanometer, and yet your brain can still be fooled into perceiving a series of 24 still images a second as one continuous motion. Now what makes you think your ears can tell the difference when the frames are moving tens of thousands of times a second?

    Even if they could, the results should be fairly predictable rather than "oh really we can tell the difference." So far, not one person has ever come forth with a list of specific types of artifaction that should be immediately evident in AAC given the introduction of specific frequencies or combinations of frequencies. They all only claim that they can hear a difference... but they can never readily quantify what that difference is.

    If that difference is itself unpredictable and not replicatable amongst the general population, then for all intents and purposes it may be the person's imagination... and they do stand a chance of being right even 30% of the time by blind guesses.

    If you could tell the difference 100% of the time with AAC and PCM in a real double-blinded, randomized, controlled setting, then something was done to master the AAC file very poorly relative to the mastering of the PCM file, or an inferior third party encoder was used in place of a readily available, superior AAC encoder that does not produce consistently flawed output... in which case one specific encoder, not the format, is at fault.

    Just out of curiosity, where do you get your 24-bit master recordings from? Or are you just stating that FLAC has the ability to play 24-bit audio but you haven't explored it. There's a reason I ask but we'll get back to that later.

    This is where your claim gets even more dubious... Did you just attempt to claim that FLAC, an off the shelf lossless format, sounds better than 16-bit Linear PCM? Normally, this is physically impossible... FLAC should not sound any better than 16-bit LPCM unless there is a difference in equipment. However, if as you suspect the cause is a difference in equipment then I have no reason to trust that your listening tests between AAC and PCM were done in an appropriate setting when your statement about FLAC versus your CD player suggests a disparity in playback equipment for one format versus the other.

    Unless you have a considerable library of 24-bit Linear PCM recordings, the results from the Squeezebox and AppleTV should be the same. Here's another area where I think you've been conned by god knows what literature into believing that the tiniest differences from one piece of digital audio hardware to another are readily perceptible which is simply false and yet another reason why I believe your claims are heavily biased.

    To put it simply... I have heard these claims before. The more an opinion is repeated does not make it fact. I am waiting for someone to actually do two things:

    a) Demonstrate mathematically, mechanically, etc. that there is a fundamentally perceptible difference between 128 Kbps AAC and 16-bit Linear PCM... i.e. be able to explain what the differences are, to show mathematically and visually where the artifaction should take place, to show that it is within the threshold of perceptibility, and

    a) Prove scientifically, with a statistically significant margin above results expected by placebo effect alone, that the difference actually is perceived consistently. In order to be scientifically relevant, the findings have to be repeatable with different source material, different sound systems, different test subjects, different settings from one trial to the next, to prove that none of these factors are responsible for the test results.

    Scientists know fairly well what the limits of human perception are, and while they seem to vary greatly from one person to the next, the distribution of hearing ability is pretty uniform across the general population (hence the usefulness of a random sample population).. that is to say that we don't have significant subsets of people whose frequency response and sensitivity falls outside the A-weighted range... just like we don't have gene pools capable of producing offspring with purple polka dotted eyes. We also can predict the incidence and frequency of artifaction of a medium if we understand how it works. Therefore, if there is such a claim as the lack of transparency between 128 Kbps AAC and 16-bit LPCM, not only should it be consistently perceptible to different people in different settings, but how and where and why it is perceptible should be consistently predictable as well... just as it is a mathematical certainty that a 7kHz sinewave sampled at 8kHz will produce an consistently perceptible 1kHz alias frequency.

    The fact that neither are true leaves the burden of proof unfulfilled in this case.
  19. bgalizio macrumors member

    Apr 28, 2006
    I'm not going to get into an arguement over scientific vs. nonscientific testing. I'm an engineer (chemical), so I very well know the merits of a peer-reviewed paper. However, we're talking personal enjoyment of music here. The double blind tests that I've participated in are simple at home listening tests. I'm not trying to argue that no scientific paper has been published yet - I agree that it hasn't.

    The system and environment absolutely should have an impact on musical differences. I'd wager that I can't tell the difference between 128 Kbps AAC vs. PCM in my car. The noise floor is high. The speakers aren't very good. But, at home, I have had no problem with this test. And I don't listen in an anechoic chamber. In fact, there are plenty of reflection issues, as I don't have a specially designed listening room. It's just part of my finished basement. No treatments, etc.

    I get my 24bit master recordings from my own gear. I am an audio recording hobbyist (aka: taper). With the advent of realtively high quality, inexpensive 24bit recording gear, it is very easy to make 24bit recordings as a hobby. Dithering issues aside, 24bit vs. 16bit is generally an easy-to-tell difference. I'm sure this could be argued scientifically, because the of the dynamic range advantages of 24bit.

    FLAC vs. CD does show a difference to me, and I'm reasonably certain it's a difference with the transport (Squeezebox vs. my CD player). I don't have any other way to explain it.

    In my double blind AAC vs. PCM tests, the transport has always stayed the same. One example: iTunes playing AAC track vs. iTunes playing PCM track (optical out to the same DAC > amp > speakers). Again, always double blind. We did this one at differing AAC bit rates. I couldn't identify which was which once we were at or over 256 kBps, but at 128 it was easy for me.

    Regarding Squeezebox vs. Apple TV - like I said, I don't own an Apple TV, nor have I listened to one outside of the Apple store. This is definitely an opinion, but differing DACs could definitely cause an audible difference. I'm talking analog out of the Squeezebox/Apple TV, not digital out to the same outboard DAC.

    All scientific papers aside, try the test yourself sometime (if you haven't already). I agree that there is plenty of snake oil in high end audio. I sure as hell better believe I hear a difference if I buy, say, $100,000 speakers vs. even $20,000 speakers! But I don't think the 128 kBps AAC vs. 16/44.1 test is one of them.
  20. MagnusVonMagnum macrumors 601


    Jun 18, 2007
    Lossless is definitely lossless. I know because I dumped my entire 360+ CD collection into lossless and I have a dozen or so DTS surround music CDs and they simply would NOT play properly (due to surround encoding DTS format) if iTunes did not spit it out bit-perfect once again. They play flawlessly on both Airport Express AND AppleTV (no real difference between the two if you're using the optical out jack on either one and an external DAC or receiver). However, I think the DAC in the AppleTV might be a bit better than the one in the Airport Express, but I can't say for certain. I know what's in the Airport Express and it's not great.

    In any case, there's NO advantage to using AIFF or WAV (iTunes supports both) over Apple Lossless. The output is identical either way. It'd be nice if Apple supported FLAC (open standard), but I have software that can convert my entire library either way if needed so I put it all in Apple Lossless so I could use iTunes + AppleTV and/or Airport Express for the simple reason that I tried out the Squeeze Center 7 software and a virtual Squeezebox and Squeeze Center is SLOW AS MOLASSES compared to iTunes for song searching type operations. Squeeze Center couldn't play Apple Lossless perfectly either (apparently the reverse engineered ALAC decoder has a few glitches to it as I'd hear a click/pop once in awhile that simply never happens with iTunes), but I'm sure I would have converted over to FLAC if I went the Squeezebox route and just kept an AAC library for my iPod Touch (need to copy/covert one anyway for it as a 16GB iPod Touch can't hold even close to my 114GB losslessly compressed library) without using AAC.

    AppleTV can do some neat slideshow presentations to your music library also. I use a 720P projector on my home theater with a 93" screen and it's fun to watch photo albums play back with all kinds of dissolves, etc. and have music playing wirelessly across the network from the Mac server I use for music/video streaming and web browsing.

Share This Page