OK - in spite of being able to ask some of the people who actually work on the BL5s, I've never been able to get a perfectly straight answer to my questions about wordlength and samplerates.

I see that a member over at the B&O Community Forum is wondering the same, so I'll repeat his question here. Who knows - for certain? This is relative to the coax s/pdif connection:

Does the BeoLab 5 accept 24-bit wordlengths, or does it downsample to 16-bit during D/A conversion?

Does the BeoLab 5 accept high resolution sample rates such as 96khz and 192khz?

The literature mentions only sample rates of 44.1, 48, and 88.2.

Where is detailed info & specs regarding the BeoLab 5 built-in speaker protection circuitry?

I can only refer to the Swedish magazine HiFi&Music.. I have a number from 2004 when they did a big review on the BL5´s with detailed specs, and the source was from B&o Denmark they wrote...

I have the magazine in my summerhouse so i can´t copy the exact detailed specs here, but i remember to 99-100%

That the D/A converter handle´s wordlength´s and samplerates up to 24Bit/96Khz with S/pdif digital signal (in 2 channel mastering).

The A/D converter converts all incoming signal to the same wordlength´s and samplerates as i wrote above (24Bit/96Khz) with Powerlink or Line signal´s.

So if you got a sample on your computer with 2 channel mastering and 24/96Khz you should be able to play it without problems..

If i remember right i think the HiFi&Musik review used a dCS Verdi La Scala CD/SACD transport player (30.000 €)

And they fed the BL5´s with up sampled signal´s from cd´s and SACD´s with samplerates up to 24/96Khz.. But when they fed the speakers with 24Bit/384Khz from the dCS Ultra High end player the Beolab 5´s was quiet...

So i can´t say to 100% if the BL5´s later in the processing down-samples the incoming 24Bit/96Khz signal to 16Bit/44.1 Khz ... It sounds most unlikely too me...

But try to download a 24Bit/96Khz or a 24bit/192Khz master sample from Linn record list and see what happens... 

Regards    

Thanks. I've hunted for official spec's, as the rates I've found are also in accordance with the message from the Community Forum quoted above.

I have run Linn Records higher length and rate files through them without problems, but if the speakers are downconverting that doesn't have to mean much.

(I can downconvert these files easily with the DAC in my Mac, for instance, using Audio MIDI Setup.)
 

Thanks for starting this subject.  The most I have ever found on this is from Ultraaudio:

The Beolab 5 is a very digital loudspeaker. While it can accept either analog (phono socket) line-level signals or direct digital (S/PDIF) connection, it immediately converts an analog input into digital code via a 24-bit/96kHz A/D converter. The DSP processing is handled by a 32-bit Sharc 2 DSP, while a 512kbyte Flash-ROM memory permits upgradeability.

Do they plan to upgrade the DSP?

The other mystery to me is why the demonstration Beolab 5s in the stores never use the digital in connection?  I was told by sales staff that people like to be able to adjust the tone of the sound or they thought it sounded better this way.  The only direct answer I got was from a B&O engineer from headquarters when I ask about using a turntable in a Beolab 5 setup.  He felt the sound would not be as good but would be close to a CD.

Sorry I don't have a definitive answer to your question.  However since I am thinking about getting Beolab 5, I find this topic to be improtant.

Thanks 

I know I am not answering your question but I wanted to add this information about HD ACC encoding that is coming.

http://www.macworld.com/article/131447/2008/01/hdaac.html 

I'm asking the question because I'm curious as to whether the BL5s will process the "new" HD sound formats properly, or whether it will downsample them.
I'd also like to be on top of the specifics as to the signal processing in the speakers, and as this thread demonstrates it's hard to get a clear-cut answer!

1. Will a 24bit/192kHz signal be processed as such, or will it be downsampled?

2. If I'm using an external DAC, and feeding the speaker with an analog, already converted signal, will that signal undergo additional processing before amplification? And if yes, to what extent/with what implications?

Let me end by saying that as of now, this is not a major problem, since most of my music is in Redbook (16bit/44.1kHz) - but I am looking forward to the HD-audio format of Blu-ray and wish to be certain that I'm setting up right to take advantage of that!
 

Yes it would down-sample the 24/192khz signal to 24Bit/96Khz  because of the limitation in the DAC clock. It can´t handle higher samplesrates than 24bit/96khz in two channel masterings as i wrote above...;) 

And if you have a separate DAC and feed the BL5´s with an analog upsampled signal it would be sampled again internally with the A/D 24Bit/96Khz converter, and give out 24bit/96Khz wordlenghts and sampelrates...

But for the best result: feed the BL5´s from a good CD-transport like dCS or TEAC with S/pidf for example... (Less A/D-D/A clock conversions in the signal chain = cleaner sound)

Regards

Dave Moulton:

II'd like to apologize to Daniel for the geekiness of this - in this thread we're really poking about for upsampled angels on the head of the digital pin!  

Apology accepted, Dave!

It´s actually fun to read your discussion. It´s like trying to follow a conversation between a group of Nobel Laureates about quantum physics.

i thought i'd do a quick test to see just how BeoLab 5's perform in regards to word length, so i went and bought the sound track to mary poppins.

My BL5's had no trouble with Supercalifragilisticexpialidocious.

Sorry i'm bored at work and i think i'm very funny.   rotflmao

oh and by the way, this is probably the geekiest of all geeky threads i've ever read on this site.

jazz

jazz:

i thought i'd do a quick test to see just how BeoLab 5's perform in regards to word length, so i went and bought the sound track to mary poppins.

My BL5's had no trouble with Supercalifragilisticexpialidocious.

Funny!  

To ungeek a bit. A word in computing and in digital media is a group of bits that are handled together by the processing architecture (a CD-player, for instance.) The size (or length) of a word describes the number of bits.

I think the first B&O CD-players were 14-bit, unless I'm mistaken. The CD-standard was later set at 16-bit, and there are now downloadable music files of 24-bit length. (While most audio engineers will be working at 32-bit by now - giving them more to play with when working with the files, without going below the 16-bit CD-standard.)

Think of the wordlength as a scoop of, in this instance, musical data - a snapshot of the sound at a particular point in time.¨

Which takes us to the sample-rate, which is how many times per second that snapshot (or sample) is taken. Again, for CDs, this is 44.100 times, usually written 44.1kHz. For DVDs it's 48.000 times. And in the newer, high definition sound formats you'll have even higher sample rates. The theory being that the more samples per second, the more organic will the music sound.

The original 44.1kHz sample rate for CDs was chosen with reference to something called the Nyquist Frequency. 44.1kHz is the sample-rate required to be able to process audio signals within the human hearing range without distortion. In hifi circles one speaks of a need to have higher sample rates, though the jury is out on whether we can actually hear a difference between for instance 44.1kHz and 96kHz, or the other, higher sample rates.

(In principle, a Nyquist frequency just larger than the signal bandwidth is sufficient to allow perfect reconstruction of the signal from the samples. However, this reconstruction requires an unrealizable filter that passes some frequencies unchanged while suppressing all others completely (commonly called a brickwall filter). When realizable filters are used, some degree of oversampling is necessary to accommodate the practical constraints on anti-aliasing filters. That is, frequencies close to the Nyquist frequency may be distorted in the sampling and reconstruction process, so the bandwidth should be kept below the Nyquist frequency by some margin that depends on the actual filters used.

For example, audio CDs have a sampling frequency of 44100 Hz. The Nyquist frequency is therefore 22050 Hz, which is an upper bound on the highest frequency the data can unambiguously represent. If the chosen anti-aliasing filter (a low-pass filter in this case) has a transition band of 2000 Hz, then the cut-off frequency should be no higher than 20050 Hz to yield a signal with negligible power at frequencies of 22050 Hz and greater.)

jazz:

My BL5's had no trouble with Supercalifragilisticexpialidocious.

I´m glad I´m not alone!