HarryHWombat

Clearly something is wrong somewhere as you have seen for yourself the independent tests showing 6MB/s write and 8MB/s read - more than enough for (probably) multiple high definition streams. I would guess it can only be either a faulty DroboShare or a faulty network. have you tried just a raw copy between two computers and see how fast this goes? Do you have an old ethernet hub somewhere which is stalling the network down to 10Mb/s ... ? The 6Mb/s you see is mighty, mighty close to what one would expect to see over a 10Mb/s network - could be a coincidence, of course, but also a mighty hint.

The Drobo - my Drobo at least - will take maybe five seconds to come out of what I assume is a semi-sleep (a nap?) state . I have never noticed any pauses or gaps when streaming although tbh I mostly stream ripped TV and movies through the AppleTV so buffering may hide this issue if it exists. @Jesus 6MB/s should be ample to stream movies ripped or otherwise. What speed network do you have and is it wired or wireless. Is it an "old" 10Mb ethernet?

@Jesus are you sure you mean bits per second and not bytes? I own a drobo so am obviously interested. Googling around "droboshare performance" show tests yielding 6MB write and 8MB read performance over gig Ethernet. This is not good for video editing but shld be fine for a 720p stream. Maybe your issue is elsewhere. Or I may have missed something.

The issue for me is the switch between wired and wireless. If it is connected wired and then on the host mac I turn the airport on the appletvs drop. I think it is a known issue.

@prosoft Not sure if this is the case with you but i can tell you the AppleTV can disappear from my iTunes (this is Mac based so may not be applicable to your situation). My AppleTVs are connected through a plug network (wired) as is the main Mac. If the main mac switches to Airport (wireless connection) the AppleTVs lose their connection to iTunes. Similarly, if one changes from wireless to wired the connection also drops. This generally happened when my kids logged into their accounts which for some reason switched to using the wireless network - or maybe they just fiddled around. I would guess changing from wired to wireless or vice versa changes the IP address of the host computer. Once dropped that seems to be that - I need to restart iTunes. So, if you have both wired and wireless networks this may be your issue - or at least it may give you some hints.

This is from the product information for the Chord 64 However, 44.1 kHz sampling can be capable of accurately resolving transients by the use of digital filtering. Digital filtering can go some way towards improving resolution without the need for higher sampling rates. However in order to do this the filters need to have infinite long tap lengths. Currently all reconstruction filters have relatively short tap lengths - the largest commercial device is only about 256 taps. It is due to this short tap length and the filter algorithm employed that generates the transient timing errors. These errors turned out to be very audible. Going from 256 taps to 1024 taps gave a massive improvement in sound quality - much smoother, more focused sound quality, with an incredibly deep and precise sound stage. and At this stage, a new type of algorithm was developed - the WTA filter. This was designed to minimise transient timing errors from the outset, thereby reducing the need for extremely long tap lengths. The WTA algorithm was a success - a 256 tap WTA filter sounded better than all other conventional filters, even with 1024 taps. WTA filters still benefit from long tap lengths; there is a large difference going from 256 taps to 1024 taps. I don't believe that transients have to be high frequency - I think they have lower frequency components but that last (natch) for only very short time periods. So they should get through the ADC filtering ... ? But are they then "smoothed" away by the filtering because that tap lengths are (according to Chord) too short? Are Chord talking marketing foo or is there some truth there wrt tap length/type and transients? This is how threads like this are extremely useful for me as they can help separate the science from the marketing foo.

No but would love to pop up to yours for a listen (along with the Weiss). I get frustrated with the lack of explanation and the bit about the servo is just plain wrong if iTunes is only muted and still accessing the disk.

It is also nonsense. As far as I can make out Amarra mutes iTunes therefore iTunes is still accesssing the disk there reducing servo noise cannot be true as it isnot under amarra control. Uness iam missing something or amarra pauses iTunes not mnutes

@Peter This seems to be exactly what I have found. The higher the frequency the less dots or samples you have the more etch-a-sketch the reconstructed wave becomes. Around 10KHz or below this seems to happen. O have a question about transients later when at a computer. @i_s I was being stupid. Quadratic will not work you need at least third order polynomial for higher frequencies as dots or samples can cover more than one maxima/minima. May try it But may run out of patience!

@i_s Very pleased to hear about the better filters This is becoming like a forum based degree course in DSP so I will allow myself the luxury of maybe one more post and then retire to a book! One point: I don't see why software based interpolation should produce pre or post ringing. With software one need not use feedback loops as I would suppose there are many other ways of doing it. Maybe I am confused though. Harry PhD DSP (distinction)

Right, so I constructed my spreadsheet based around @i_s - it may be right it may be wrong but the results looked good enough to tell me it was probably ok. I found that as the frequency of the input sine wave increased the error between the interpolated point and the "real" point sin (2*pi()*frequency*time) grew to be about 5%. At lower frequencies it was about 1%. Going on from @Max excellent post it seems true that we cannot accurately recreate a 20KHz wave (neither can we accurately recreate a 10KHz wave but we are closer) - this is obvious, I guess, but is news to me. (Maybe the analogue filter gets us closer, dont know). So to rephrase my previous question - could we do a better job of recreating the wave using a software pre-filter on the computer side. We could take as long as we wanted to to do this to get better results - use more computationally expensive interpolation models - we have more time (as much as we care to spend) and a faster processor. I have some sort of memory that Weiss have software to do this - is this true? Would it make sense to have a "real-time" software pre-filter to over/upsample using better interpolation before the signal exit the computer. Also - I am going to play around with quadratic based interpolation around three real samples to see if this gives closer/better results.

@i_s I get it now. Very interesting. Only really clicked when I did my own spreadsheet. Have some questions/observations for later if you don't mind (when not on iPhone). @max so is it true that theoretically one can accurAtely reconstruct a 20 wave using 40 sampling practically in a dac one cannot. More to ask here but basically would software pre upsampling give better results?

Still working on understanding how the filter works. Closer now.

I had a look at the Sheet2 on your previous sheet (after copying it I can see the formulae) but is quite difficult to what is actually going on. So maybe some questions (I couldn't work out the answers to these form the wikipedia page which I thought may help). The weights in the filter - how are these related to sinc() and to the Kronecker delta function. Is the weight zero if the sample is equal to the previous sample (ie the sample is effectively discarded)

Haven't had time to ... err ... adequately filter all this information but thought I'd bump the thread 'cos I think it deserves it One question though: what we seem to do here is to created a stepped wave and then filter this wave. Mathematically it should be possible to create a sine wave from the sample points. As one assumes higher frequencies are pre-filtered at the AD stage we should be able to take three(?) points and interpolate a sine between them as we know that that line is monotonically increasing, decreasing or turning the corner.. (Maybe it is more than three points). Is this rubbish or just not possible with electronics or do one of the existing filters perform the same funciton but in a different way. To put more simply - is it not possible to directly re-create the wave so no filtering is required?