Re: [SlimDevices: Audiophiles] Popping noises with Transporter firmware 81 and 85
audiomuze;600566 Wrote: > Both Koln Concert and Raven studio masters compressed at -8 play back > without any clicks/ pops, even with the VU meters going. But is the VU meters the most CPU intensive visualizer? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=84133 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Popping noises with Transporter firmware 81 and 85
It is my understanding that CPU load is highly dependent on the visualizer in use. Andy, what visualizer results in highest probability of audio dropout? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=84133 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
[SlimDevices: Audiophiles] T+A Music Player
Any thoughts on this one? http://www.taelektroakustik.de/eng/index.htm In the same price range as Transporter... -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=40119 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Why does it sound better?
Sean, these are impressive plots. Can you share with us some details on how you have designed the jitter suppression when receiving digital input from an external source? (I understand from your plots and explanation that you are not using asynchronous sample rate conversion.) Also, how is the word clock output mode working? There is no dedicated word clock output connector. Are the word clock signal instead sent to all the SPDIF/AES output connectors? Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=28126 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Transporter, Inside Out
seanadams;141123 Wrote: > This is the best explanation I've found: > > http://www.tnt-audio.com/clinica/diginterf1_e.html Thanks, but what connection/component scenario will most likely use Transporter's word clock input? Is the intended application of the word clock input to use Transporter with an overly expensive external dac with word clock output that might be better than Transporter's internal dac? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=27927 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Transporter, Inside Out
seanadams;141094 Wrote: > That is correct. Word clock in should absolutely not be used to drive > the internal DAC. That is not what it's for. What is the main application of word clock input? Connection to an external dac with word clock output, or connection to a digital equalizer for room acoustics correction (about which I have no knowledge)? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=27927 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Transporter knob is very cool
Is the knob an off-the-shelf component? If so, do you have a link to a datasheet or similar information? Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=27981 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: 1K Balanced impedance
seanadams;136660 Wrote: > That's basically it, with the further point I just thought of: adding > passive attenuation would be a bad idea unless the source Z in low. > Really it looks like 100Ω is the way to go - since nobody's > rooting for 1K, I'm going to change it. Will this change be applied also for the first production batch? Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=27445 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: prove squeezebox doesn't upsample?
chaotic33;132420 Wrote: > I have been using the file diatonis_dts_tatu.wav which you can find on > the internet to test if my pc configuration with kernel streaming was > not upsampling on my dts capable receiver. > If I play this file on my new squeezebox it plays static. I know that > this isn't a full proof test but is there one? Are we sure the > squeezebox isn't upsampling? I believe your question is if squeezebox outputs bit-perfect data on the digital outputs. Squeezebox2/3 indeed does if you ensure to disable the volume control for the digital out. The same applies to Squeezebox1/G, but the sign of the samples are inverted. Search for 'bit-perfect' or 'bit-correct' to find threads discussing this. Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26891 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Stereophile review
mauidan Wrote: > Did JA do jitter measurements? Yes, and he was very pleased with it. He ended the measurement section with: '... the excellent jitter rejection came as welcome surprise.' -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26306 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Stereophile review
Just got my digital September issue of Stereophile. It has the review of Squeezebox. JA concludes 'Very highly recommended'. Well done! -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26306 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: word clock output
seanadams Wrote: > No. A DAC should have a clock _output_ (whether word, mclk, or other), > not an input. You want the clock source in the DAC. > > Have a look at this: > > http://www.tnt-audio.com/clinica/diginterf2_e.html > > You should read the whole thing, but what we're talking about here is > the "Clock backwards configuration" - this is the ideal. > > -"Now, transferring the clock from another unit through a couple of > pulse transformers, two couples of connectors, a cable subject to RF > interferences and ground loops, is definitely not the best way to keep > it clean... > > The solution proposed by a few companies (Wadia and Sonic Frontiers, > between the others) is to move the master clock where the cleanest > clock is needed: that is, in the DAC unit. "- What is the difference between the terms 'master clock' and 'word clock'? Is the frequency of a word clock equal to the sample rate, while the frequency of a master clock is equal to a (possibly large) multiple of the sample rate? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26202 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: word clock output
ackcheng Wrote: > Thanks for you answer. But that's get me even more confused. I thought > the use of an external clock source i.e. master clock is to synchronise > all the digital equiptments? So the word clock output of the master > clock is connected to the word clock incput of various digital > devices. > > Isn't this arrangment is supposed to be the best? This master clock should be the DAC clock for minimum jitter. Best setup: DAC with wordclock output connected to wordclock input of the digital sources. -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26202 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: word clock output
seanadams Wrote: > No, not if a word clock is used. > > The effect of the word clock is to change the s/pdif interface from > sending "timing plus data" to just sending data. Timing then becomes > the responsibility of the receiving unit, and jitter performance is > determined only by the DAC unit's internal clock. > > It also means that as long as all the bits make it across correctly, > you don't have to worry about the quality of your cabling. If it's not a too expensive addition, then it would be nice to have word clock input on your upcoming SB4 :) -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26202 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: word clock output
seanadams Wrote: > True, there are a lot of ways to potentially slice it into two boxes. > But the costs don't work the way you think - a good chunk of it is the > chassis itsef, and there are many other cost savings as well as > functional advantages to having everything in one box. > > That said, Transporter's processor, wireless, and front panel hardware > are modular and could potentialy be upgraded separatey from the audio > section. However in reality it is extremely difficult to predict what > specific hardware lies ahead. While we did once offer a very popuar > hardware upgrade kit (the 280x16 dispay for SB1) it's not like we can > say yes there will be a CPU upgrade next year or anything like that. > Obviously we recognize that Transporter is a significant investment for > many, and we've designed it to have a very long useful life by making > the hardware extraordinarily flexible. For example, the word clock > output feature was a trivial programmable logic change - no hardware > change necessary... and there are a number of other such features on > the drawing board to take advantage of transporter's IO capabilities in > interesting ways. That is the sort of flexibility we have. Ok, I fully understand your reasoning. But would the sound quality be compromised with a two-device system having word clock communication? I guess the jitter performance could potentially be affected? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26202 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: word clock output
seanadams Wrote: > I was doing some measurements recently on Transporter, using the dScope > analyzer playing through Transporter in "standalone DAC" mode. One of > the things I thought could be holding us back was jitter from the > analyzer's clock, and on the s/pdif connection between the analyzer and > Transporter. > > So I added a feature to send a word clock output signal on our s/pdif > ports, to which the analyzer can then "slave" its clock. Below is a > plot of a 8KHz sine wave with and without word clock sync. Neat eh? > > The jitter from the analyzer was not even particularly bad compared to > many devices I've measured - around 200ps RMS. But there is quite a > difference compared to Transporter's internal clock! > > So if you have a CD player or other source that can take a word clock > input, you will be able to hook it up like this by checking an option > to use word sync when in DAC mode. Let's imagine a SB3 with a wordclock input, that has the digital output connected to the digital input of the Transporter. How would the performance of such a setup compare to the quality of the Transporter? The reason for asking this question, is that as I see it, the Transporter consists mainly of two parts. The first part is somehow on the "computer side" of the device. This part is responsible for the network communication with slimserver, the display, and so on. The other main part is on the "audiophile side" of the device. This second part consists of the DAC, the analog stages and so on. Computer technology and standards seem to me to evovle more rapidly than audiophile technology. For instance, the Transporter might be considered "old technolgy" when 802.11n becomes the wireless network standard. This would be a pity because the DAC etc. would probably still be state of the art. Would the sound quality have suffered if the Transporter was made as two different components? One 500$ SB3 type device with wordclock input, larger display and buttons on the front. It should also be possible to operate this device alone with sound quality at SB3 level. The other device would be a 1500$ DAC with wordclock output and state of the art audio circuitry. The two devices together would bring networked music to the audiophile level. Any thoughts on this? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=26202 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Transporter design question
325xi Wrote: > > Question: Transporter has digital IN. Could you describe what jitter > reduction mechanisms are involved when datastream comes from an > external source? seanadams Wrote: > > We do what we can (wrt power regulation and signal routing) to not make > it any worse once the signal is received, but obviously it can never be > as good as a local clock source. For the S/PDIF input, is there a PLL, or is the clock retrieval based on asynchronous sample rate conversion (ala Benchmark)? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=25919 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Slim Devices Transporter?
What about headphone outputs? Can't find any information about this on the web page... -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=25773 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Is FLAC bit perfect decoded on SB3
dlite Wrote: > When a SB3 decodes a FLAC (or other lossless format) how is it done? Is > the FLAC decoded to a wav file before being played? How do i know it is > bit perfect? Is there anyway to take the digital output from the Digital > out on the SB3 and write it to a file to compare against the original > wav file? > > Also when the flac file is transmitted either wirelessly or via > ethernet cable, do network protocols ensure the file is bit perfect > when it reaches the SB3? Take a look at http://www6.head-fi.org/forums/showthread.php?p=1297085 -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=23174 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Softsqueeze and bit-correct output on Windows
banzai Wrote: > SoftSqueeze is running on my Win2k box using the MAudio 24/96's ASIO > driver. > > I get occasional pauses - as if there were network congestion (unlikely > given my gigabit LAN) - but otherwise it seems to work just fine. OK, thanks. You do this by selecting the ASIO driver (instead of the 'Primary Sound Driver') in the Audio mixer setting in Softsqueeze Preferences? -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=22636 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Softsqueeze and bit-correct output on Windows
dwc Wrote: > Steinar, > Bit-correctness depends on the sound card hardware. You need to buy a > card with a 44.1 crystal on it to maintain 44.1 output. Most cheap > cards and onboard sound only have a 48 crystal so they must resample > (even if you run kernel streaming or asio). > > The cheapest bit-perfect (for 44.1, use kernel streaming) soundcard > that I've read about is the chaintek av710. > > Here's an example setup guide for it. > http://www6.head-fi.org/forums/showthread.php?t=98588&highlight=chaintech+guide But will Softsqueeze work with kernel streaming or ASIO drivers? I guess what I am asking is if special functionality needs to be included in the application for it to work with these sort of "non-standard" Windows audio interfaces. -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=22636 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Softsqueeze and bit-correct output on Windows
MrStan Wrote: > In order to stand a chance of "bit correctness" I suspect your soundcard > would require a digital mixer. The problem is that for a digital mixer > to work all inputs have to be resampled to the same rate and clock to > synchronise. This could well kill any chance of "bit correctness" > although it would sound fine. I believe that what is needed is for the mixer NOT to resample, i.e. keep CD audio data at 44.1kHz. -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=22636 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Softsqueeze and bit-correct output on Windows
banzai Wrote: > In theory you're right, though in practice who knows? You're still > counting on your soundcard driver/manufacturer to pass the signal > through cleanly. > > I've got my living room PC running an ASIO driver through my MAudio > 24/96 card. It sounds exceptional, but I haven't tried to compare a > FLAC/WAV file via the PC vs my CD transport. > > One thing that gave me pause was that I think my sound card's mixer > (I'm not talking about Windows volume control) allows me to attenuate > the outgoing signal. I wasn't happy to see that. The fact that the > mixer is willing to change the signal isn't a good sign. Perhaps at > zero attenuation it is a bit-perfect transfer. But again, who knows? One way to find out if the digital out is bit-perfect is to play a wav encoded surround file, and connect the digital output of the sound card to the digital input on a surround receiver. The surround sound will not play back correctly if the bit stream is not perfect. Wav encoded surround files can be downloaded from http://www.sr.se/multikanal/english/e_index.stm I don't have proper hardware at the moment to do the test myself, but I am interested in this because I plan to upgrade my soundcard... (When doing such a test, ensure that the volume on your receiver is turned down because if the output is not bit-perfect the resulting output from your receiver might be extremely loud noise.) -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=22636 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Softsqueeze and bit-correct output on Windows
Does anyone have any knowledge about bit-correctness of the digital output from the soundcard when running Softsqueeze on a PC running Windows XP? It is my understanding that the conventional Windows audio functionality will do resampling of the audio stream thereby destroying the bit-correctness of the audio stream sent to the digital out. It is also my understanding that an ASIO enabled soundcard will bypass all Windows audio mixing, and thereby preserving bit-correctness. Will Softsqueeze on Windows XP together with an ASIO soundcard be a bit-perfect "transport" for an external DAC? (Not considering jitter etc, just the bit pattern) Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=22636 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Volume bug - can you really hear it?
Andrew L. Weekes Wrote: > I believe you'll find the changeover is at 16 on the 0-40 range. That's > what the measurements indicate - below this value the old and patched > code measure identically. > > Andy. The first checkin of the patch had the changeover at -35dB, this corresponds to a volume setting of 12. The final version of the patch changed the changeover to -30dB. As you correctly states, -30dB corresponds to 16 on the 0-40 range. Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=18439 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Volume bug - can you really hear it?
cbemoore Wrote: > Hi Dean, > > Your change applies 8 bit accuracy to volumes louder than -35dB. What > volume on the SB 0-40 volume scale corresponds to -35dB? > > Chris I believe the volume changes in steps of 1.25dB. That means that -35dB corresponds to a volume setting of 12 in the SB 0-40 volume range. Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=18439 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Volume bug - can you really hear it?
seanadams Wrote: > Having dismantled a DAC-1 I can tell you it is a perfectly vanilla > implementation of ASRC using a standard ADI chip, and it does perform > as advertised WRT to jitter rejection - this feature is _inherent_ in > asynchronous sample rate conversion. > > ASRC effectively moves the source of jitter from the s/pdif input DAC's > internal clock. That's fine but it's not necessarily an improvement - > the latest standard s/pdif receiver chips from Crystal, ADI, TI all > have very good jitter attenuation - in fact a good PLL design will > inherently act as filter to clean up the recovered clock however bad it > is. > > However, ASRC while it obviously eliminates jitter in the source > signal, does god knows what to the data coming through. > > That said, the Benchmark's DAC and amplification stages appear to be > carefully engineered, and measured performance here is great. > Unfortunately real performance is harder to market, but ELIMINATES > JITTER sells. ;) I found this tutorial on asynchronous sample rate conversion: http://www.diyaudio.com/forums/showthread.php?threadid=28814 Might be of interest to you guys. Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=18439 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Volume bug - can you really hear it?
seanadams Wrote: > Aside from the volume levels being different, I can't hear any > difference between the old firmware and the new firmware. We are going > to put the patch in to use 8-bit coefficients BUT we are concerned that > if there is really an audible difference that it may be some other bug. > I skeptical of the idea that rounding in the 24th bit would be audible, > but I'm quite open to the possibility that there's something else going > on. > > It would be good to get real confirmation through blind testing or > measurements that: > > 1) there was really a problem introduced after FW 15 > 2) the problem is or is not fixed by this patch > > There are many people here with better ears than mine - if you can > really hear it then we need your help to identify the problem and > confirm a fix. Sean, you previously suggested to record the digital output in order to verify the correctness of the multiplier doing the volume adjustment. By doing that you will avoid a lot of speculations whether there is a bug or not. I think it is worth to invest the time necessary to do such a test... Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=18439 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: How to patch SS code for volume control bug?
pfarrell Wrote: > On Sun, 2005-11-20 at 09:15 -0800, sbjaerum wrote: > > This will of course only work if you are running the perl version of > > slimserver, and not the compiled Windows binary. > > I'm not sure that this is true. I thought that the "windows binary" > was > just perl and the perl source needed to run. Perl really isn't > a compiled language. > > > -- > Pat > http://www.pfarrell.com/music/slimserver/slimsoftware.html My understanding is that the windows .exe version is built using the Activestate Perl Dev Kit. That means that you do not have to have the Perl interpreter installed to run slimserver on Windows. To apply patches you will have to rebuild. Running the perl version of slimserver means that patches in perl source code does not need a rebuild. Someone with more knowledge in perl than me, should probably explain how the .exe is built using the Perl Dev Kit. Steinar -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=18369 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: How to patch SS code for volume control bug?
Patrick Dixon Wrote:
> Edit the file Squeezebox2.pm in the sub-directory Slim/Player/ to change
> the section:
>
> sub dBToFixed {
> my $db = shift;
>
> # Map a floating point dB value to a 16.16 fixed point value
> to
> # send as a new style volume to SB2 (FW 22+).
> my $floatmult = 10 ** ($db/20);
> return int(($floatmult * (1 << 16)) + 0.5);
> }
>
>
> to:
>
> sub dBToFixed {
> my $db = shift;
>
> # Map a floating point dB value to a 16.16 fixed point value
> to
> # send as a new style volume to SB2 (FW 22+).
> my $floatmult = 10 ** ($db/20);
> if ($db >= -30 && $db <= 0) {
> return int($floatmult * (1 << 8) + 0.5) * (1 << 8);
> }
> else {
> return int(($floatmult * (1 << 16)) + 0.5);
> }
> }
>
>
> Then restart Slimserver.
This will of course only work if you are running the perl version of
slimserver, and not the compiled Windows binary.
Steinar
--
sbjaerum
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[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
Dean Blackketter asked: > Another proposal moved the 8 to 16 bit threshold from -35db to -30dB. What's the right value for this? The reason why I suggested to stop the modified gain calculations at -30dB is that below this value repeated values start to appear. In the list below, the patch is used all the way down to -50dB, and you can see when repeated values show up. Steinar dB | Trunk | Patch 0 | 65536 | 65536 -1 | 58409 | 58368 -2 | 52057 | 51968 -3 | 46396 | 46336 -4 | 41350 | 41472 -5 | 36854 | 36864 -6 | 32846 | 32768 -7 | 29274 | 29184 -8 | 26090 | 26112 -9 | 23253 | 23296 -10 | 20724 | 20736 -11 | 18471 | 18432 -12 | 16462 | 16384 -13 | 14672 | 14592 -14 | 13076 | 13056 -15 | 11654 | 11776 -16 | 10387 | 10496 -17 | 9257 | 9216 -18 | 8250 | 8192 -19 | 7353 | 7424 -20 | 6554 | 6656 -21 | 5841 | 5888 -22 | 5206 | 5120 -23 | 4640 | 4608 -24 | 4135 | 4096 -25 | 3685 | 3584 -26 | 3285 | 3328 -27 | 2927 | 2816 -28 | 2609 | 2560 -29 | 2325 | 2304 -30 | 2072 | 2048 -31 | 1847 | 1792 -32 | 1646 | 1536 -33 | 1467 | 1536 -34 | 1308 | 1280 -35 | 1165 | 1280 -36 | 1039 | 1024 -37 | 926 | 1024 -38 | 825 | 768 -39 | 735 | 768 -40 | 655 | 768 -41 | 584 | 512 -42 | 521 | 512 -43 | 464 | 512 -44 | 414 | 512 -45 | 369 | 256 -46 | 328 | 256 -47 | 293 | 256 -48 | 261 | 256 -49 | 233 | 256 -50 | 207 | 256 -- sbjaerum sbjaerum's Profile: http://forums.slimdevices.com/member.php?userid=237 View this thread: http://forums.slimdevices.com/showthread.php?t=17269 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
dean Wrote:
> I'm not entirely sure I understand the problem well enough to say
> that this patch is correct.
>
> (I realize that there has been some listening tests done and you like
>
> it better, but I want to also make sure it's correct.)
>
> Given that the Squeezebox has a 24 bit output, the rounding error
> should be down at around at 138-144dB, not 90-96dB and I have a hard
> time believing that the effect would be as obvious as you state at
> that level.
I am not sure how audible the difference is, but in my opinion it is
"cleaner" to multiply with gain coefficients that does not lead to bits
being lost because of rounding.
I suggest to modify the gain coefficients for volume settings from
-30dB to 0dB so that the patch becomes:
Index: server/Slim/Player/Squeezebox2.pm
===
--- server/Slim/Player/Squeezebox2.pm (revision 5190)
+++ server/Slim/Player/Squeezebox2.pm (working copy)
@@ -272,7 +272,12 @@
# Map a floating point dB value to a 16.16 fixed point value
to
# send as a new style volume to SB2 (FW 22+).
my $floatmult = 10 ** ($db/20);
- return int(($floatmult * (1 << 16)) + 0.5);
+ if ($db >= -30 && $db <= 0) {
+ return int($floatmult * (1 << 8) + 0.5) * (1 << 8);
+ }
+ else {
+ return int(($floatmult * (1 << 16)) + 0.5);
+ }
}
sub volume {
With this patch the differences between "trunk" gain and "patch" gain
are listed below.
The differences are small, but again, I think the patch gain
coefficients are "cleaner".
In my opinion, it can't hurt to use the patch gain.
The purists will be satisfied if it is applied...
Steinar
dB Trunk Patch
0 65536 65536
-1 58409 58368
-2 52057 51968
-3 46396 46336
-4 41350 41472
-5 36854 36864
-6 32846 32768
-7 29274 29184
-8 26090 26112
-9 23253 23296
-10 20724 20736
-11 18471 18432
-12 16462 16384
-13 14672 14592
-14 13076 13056
-15 11654 11776
-16 10387 10496
-1792579216
-1882508192
-1973537424
-2065546656
-2158415888
-2252065120
-2346404608
-2441354096
-2536853584
-2632853328
-2729272816
-2826092560
-2923252304
-3020722048
-3118471847
-3216461646
-3314671467
-3413081308
-3511651165
-3610391039
-37 926 926
-38 825 825
-39 735 735
-40 655 655
-41 584 584
-42 521 521
-43 464 464
-44 414 414
-45 369 369
-46 328 328
-47 293 293
-48 261 261
-49 233 233
-50 207 207
--
sbjaerum
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[SlimDevices: Audiophiles] Re: DAC technical question
jhwilliams Wrote: > Well, theoretically it's the same - however the noise floor is still the > same so you'll have a lower snr. The quantization noise scales with the same factor as the signal, so the signal to quantization noise is unaltered as far as I can see. But the thermal noise and other "analog" noise sources remains at the same level. This noise level then increases relative to the signal. Is this a correct interpretation? Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] DAC technical question
This is a general question, not specifically related to squeezebox. Lets assume we have a 24 bit DAC and use it to convert 16 bit CD audio data. In normal operation the 16 bit samples are placed in the 16 most significant bits of the 24 bit DAC input word. Now, lets assume that the 16 bit samples are right-shifted within the 24 bit DAC word. In the extreme case it can be shifted 8 bits to the right and still no bits are lost. The output level of the DAC will be reduced as a result of the digital scaling, but other than that how will such scaling affect the final analog signal? Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
seanadams Wrote:
> If someone has the time it would be REALLY helpful to get verification
> that the new multiplier is correct. Make an AIFF that loops through a
> few hundred handom numbers, then play it on the old firmware and then
> on the new firmware with this patch. Record the s/pdif output and
> compare. I'd do it right now but I don't have the time just this
> moment. Thanks.
sbjaerum Wrote:
> I don't think it is that straight forward. I'm not sure if there is any
> volume setting that multiplies with the same number in the old and new
> firmware.
>
> The old firmware multiplies with a gain coefficient in 1.7 fixed-point
> format. The proposed patch multiplies with a gain that is effectively in
> 1.8 format.
>
> There is thus a need for a new patch of Squeezebox2.pm before such an
> experiment can be performed.
>
> Steinar
For verification of new vs. old firmware gain multiplication, I think
the below patch would be useful. With this patch the same gain
coefficient should be applied to the data both in old and new firmware.
However, the firmware implementation is different, and that is what
should be tested.
Unfortunately I have neither the equipment nor the time at the moment
to do this. Hopefully someone else have...
Steinar
Index: server/Slim/Player/Squeezebox2.pm
===
--- server/Slim/Player/Squeezebox2.pm (revision 4951)
+++ server/Slim/Player/Squeezebox2.pm (working copy)
@@ -272,7 +272,12 @@
# Map a floating point dB value to a 16.16 fixed point value
to
# send as a new style volume to SB2 (FW 22+).
my $floatmult = 10 ** ($db/20);
- return int(($floatmult * (1 << 16)) + 0.5);
+ if ($db >= -35 && $db <= 0) {
+ return int($floatmult * (1 << 8) + 0.5) * (1 << 8);
+ }
+ else {
+ return int(($floatmult * (1 << 16)) + 0.5);
+ }
}
sub volume {
@@ -286,17 +291,8 @@
# Old style volume:
my $oldGain = $volume_map[int($volume)];
- my $newGain;
- if ($volume == 0) {
- $newGain = 0;
- }
- else {
- # With new style volume, let's try -49.5dB as
the lowest
- # value.
- my $db = ($volume - 100)/2;
- $newGain = dBToFixed($db);
- }
-
+ my $newGain = $oldGain << 9;
+
my $data = pack('NNCCNN', $oldGain, $oldGain,
$client->prefGet("
digitalVolumeControl"), $preamp, $newGain, $newGain);
$client->sendFrame('audg', \$data);
}
--
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[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
seanadams Wrote: > > If someone has the time it would be REALLY helpful to get verification > that the new multiplier is correct. Make an AIFF that loops through a > few hundred handom numbers, then play it on the old firmware and then > on the new firmware with this patch. Record the s/pdif output and > compare. I'd do it right now but I don't have the time just this > moment. Thanks. I don't think it is that straight forward. I'm not sure if there is any volume setting that multiplies with the same number in the old and new firmware. The old firmware multiplies with a gain coefficient in 1.7 fixed-point format. The proposed patch multiplies with a gain that is effectively in 1.8 format. There is thus a need for a new patch of Squeezebox2.pm before such an experiment can be performed. Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
Patrick Dixon Wrote:
> Steinar, I have just had the Rolling Stones around for tea - so I think
> your patch is good! Thank you very much. Can you tell me what the
> perl-speak is for -35 <= $dB < 0 and I will run with that.
The below patch does that.
Steinar
Index: server/Slim/Player/Squeezebox2.pm
===
--- server/Slim/Player/Squeezebox2.pm (revision 4942)
+++ server/Slim/Player/Squeezebox2.pm (working copy)
@@ -272,7 +272,12 @@
# Map a floating point dB value to a 16.16 fixed point value
to
# send as a new style volume to SB2 (FW 22+).
my $floatmult = 10 ** ($db/20);
- return int(($floatmult * (1 << 16)) + 0.5);
+if ($db >= -35 && $db <= 0) {
+return int($floatmult * (1 << 8) + 0.5) * (1 << 8);
+}
+else {
+return int(($floatmult * (1 << 16)) + 0.5);
+}
}
sub volume {
--
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[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
I think Patrick's point is that by selecting gain coefficients that are multiples of 1/256, no bits are thrown away when truncating to 24 bits. I think my patch above ensures that after multiplication, no bits need to be thrown away when truncating to 24 bits. With these coefficients, the gain multiplication is reversible, the original samples can be recovered because no bits were thrown away. Steinar dean Wrote: > On Oct 30, 2005, at 1:39 AM, Patrick Dixon wrote: > > > > > > >> It's not clear that there is an issue at all. We'd need a > >> reproducible case. > > > There is definitely an issue, and it's easy to demonstrate/reproduce > - > > the only question is whether you can hear it or not! I say I can, > and > > jhwilliams has done blind testing which indicates that he and his > > girlfriend can too. I have now reverted to Firmware 15, which I'm > > happy with sound-wise. > > Well, internally on the player, the old volume control and the new > volume control both change the same gain coefficient. > > The old-style is a 1.7 fixed-point encoding and the new one is a > 16.16. So, depending on the value of the gain, you may find rounding > > errors, since the resultant value is truncated at 24-bits, but the > trade off is that you can have much smaller volume increments. > > I can't argue that you can't hear the difference (folly in this > forum), but I do believe that it's mathematically correct and for the > > vast majority of users the new volume curve is a benefit. > > p.s. I found this article on dither very interesting: stereophile.com/features/705dither/> > > > > > > > > The issue is that previously the volume control used an 8-bit > > multiplier coefficient which meant that the full resolution of > 16-bit > > audio (which is of course what CDs are) was always retained within > the > > 24-bit datapath of the SB2. The new log volume control uses a 16x16 > > bit multiplier which means that rounding errors are 'always' > > introduced, and these are unconcealed by the addition of any dither. > > It's surprising to me too that I can hear these artifacts, but I > think > > it's the character of the noise (coupled with oversampling in the > > DAC?) > > that makes it stand out. > > > > Even with the addition of dither (which may well be useful for the > > replay gain situation anyway), there is an audiophile arguement that > > says you should always preseve the S/N ratio of the original 16 bit > > signal where you can. Therefore I am simply suggesting that you > > 'tweak' the coefficients of volume control settings 20-39 to use > 8-bit > > resolution rather using the full 16-bit resolution of the new > > multiplier. Much below vol 20, 8-bit coefficients are not enough to > > resolve the volume steps required, but at these levels the artifacts > > are unlikely to be an issue. > > > > My calculations gave something like this below - so the volume > changes > > would be negligeable with the adjusted coeffs. Just ANDing the > 20-39 > > coeffs with 1100hex would probably give acceptable results. > > > > vol8-bit coeff'new' dB'old' dB > > > > 2014-25.2-25.4 > > 2116-24.1-24.1 > > 2218-23.1-22.8 > > 2321-21.7-21.6 > > 2425-20.2-20.3 > > 2529-18.9-19.0 > > 2633-17.8-17.8 > > 2738-16.6-16.5 > > 2844-15.3-15.2 > > 2951-14.0-14.0 > > 3059-12.7-12.7 > > 3169-11.4-11.4 > > 3280-10.1-10.2 > > 3392-8.9-8.9 > > 34107-7.6-7.6 > > 35123-6.4-6.3 > > 36143-5.1-5.1 > > 37165-3.8-3.8 > > 38191-2.5-2.5 > > 39221-1.3-1.3 > > 402560.00.0 > > > > > > -- > > Patrick Dixon > > > > www.at-view.co.uk > > -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
jhwilliams Wrote: > Let me know regarding the perl you are looking at - however, I'm running > 6.2 w/ firmware 15 so it seems more likely to be a firmware side issue > (afaik?). > > Jon The perl patch works together with firmware versions with the new volume functionality. I don't remember at which version this went into the firmware. Allthough the multiplication is done in the client, the gain-coefficient to multiply the samples with is calculated by the server. The patch tries to provide coefficients that avoids round-off problems in the multiplication. If coefficients do not turn out to be the problem, there is a possibility that there is a bug in the implementation of the fixed-point multiplication in the firmware. But this is of course only speculations... Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
Patrick Dixon Wrote:
> It probably should be -whatever-the-perl-syntax-is-{-25 <= $dB < 0} (we
> could make this -35 I think), otherwise the routine will affect replay
> gain too (if it's used there).
OK, agree. Any listening tests yet?
Steinar
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[SlimDevices: Audiophiles] Re: Sound Quality w/ Latest Firmware
Patrick Dixon Wrote:
> So Sean, I have a suggestion. The log volume control is clearly a
> better approach overall, but could you implement a version that uses
> 1/256 rounded multipliers for (say) volumes 20-40, and non-rounded
> multipliers below? No critical listening can really take place at
> -25dB+ with 'normal' amp gains /speaker efficiency, so the levels below
> 20 (which can't be smoothly rounded in 1/256ths) don't really matter.
>
> In the interests of bit-perfect audio and all that
seanadams Wrote:
> Sounds reasonable, but I'm not keen on diving in right this moment to
> change it - we'll look into it though.
I don't know how the multiplication with gain is done in the firmware,
but wouldn't the small server patch below do what Patrick suggests?
Steinar
Index: server/Slim/Player/Squeezebox2.pm
===
--- server/Slim/Player/Squeezebox2.pm (revision 4942)
+++ server/Slim/Player/Squeezebox2.pm (working copy)
@@ -272,7 +272,12 @@
# Map a floating point dB value to a 16.16 fixed point value
to
# send as a new style volume to SB2 (FW 22+).
my $floatmult = 10 ** ($db/20);
- return int(($floatmult * (1 << 16)) + 0.5);
+ if ($db >= -25) {
+ return int($floatmult * (1 << 8) + 0.5) * (1 << 8);
+ }
+ else {
+ return int(($floatmult * (1 << 16)) + 0.5);
+ }
}
--
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[SlimDevices: Audiophiles] Re: DTS file for tests ?
oreillymj Wrote: > BTW - just for the hell of it last night, I thought I'd stream the > Kelly's Industry > http://www.kellyindustries.com/downloads/dts-44k_diatonis_soal.zip DTS > file to my SB2 as a flac file. > > There were a few surprising results. > > 1) The WAV only compressed from 59mb to 56mb when converted to flac. I > was expecting a much better compression ratio, based on a post by Sean > earlier saying that DTS files were padded with 0's. I was expected to > shrink the file by maybe 40-50%. BTW- the flac compression was set to > -8 (highest compression) > > 2) The flac file would not stream to the SB2 without breakup. I'm not > sure what the problem is, but even though the reception of my 802.11g > network is 80-90% (and the SB2 was the only Wireless device on it), the > SB2 did not seem to be able to fill it's buffer quickly enough to keep > the music from breaking up. When I looked at the buffer fullness > indicator, it seemed that when the SB2 was paused , the buffer would > fill to 94%, then when I unpaused, the indicator dropped in steps of > 10% until the audio started to break up. > > Someone mentioned in another thread that the SB2 had 64mb RAM on board, > with 32mb used for buffering. If that's correct,I would have expected to > get at least 1/2 way into the file before breakup. The reality was that > I only managed 10-15 seconds at a time. Did you compress to flac on the fly? If so, could the high compression level (-8) cause your CPU to be the bottleneck? Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Audiophile perspective on the new Squeezebox
seanadams Wrote: > SB3 has 99.9% the same circuitry and measures identically to SB2 on the > standard DAC tests (THD, noise floor, etc). > > The change to use a linear reg for the 3.3 rail improves jitter > measurements, but it's already so close to measurable limits for the > equipment I have that it's getting hard to quantify. Maybe if we sell > enough SB3s I can get one of those high-end scopes with the jitter > analysis option. :) Hopefully you will get a contribution from me soon :) Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Audiophile perspective on the new Squeezebox
PhilNYC Wrote: > So based on all that, the question remains...is there significant reason > for audiophiles to upgrade to the new box? eg. is there a significant > improvement in the jitter spec on the digital out? I would like to ask the opposite question. Are there any audiophile reasons not to upgrade? Are the same high-quality components used in version 3 or are different components used in order to reduce manufacturing costs? On the web pages it is only stated that v3 'is functionally identical to Squeezebox2'... Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SB2: reduce analog out voltage by 10x?
seanadams Wrote: > The feature is not in yet, but I did get as far as testing the disabling > of the DAC to confirm that this will probably work for your situation Is there a bug registered for this feature? Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: DTS file for tests ?
smst Wrote: > Do you mean that, for each 16-bit sample, the top bit needs to be > flipped? No, change 10 to -10 etc. I guess the wav samples are stored as 2-complement signed integers, so this is not equal to flipping only the MSB bit. Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: DTS file for tests ?
smst Wrote: > Update on my conversion utility: > > I've written a Python script which can convert an AC3 file to a WAV > file suitable for streaming to an SB2. My recommendation is still that > an AC3 file is pre-converted to WAV, and then to FLAC so that metadata > can be set. However, my utility does estimate the output WAV file > length (correctly, in my tests so far) so it should be easy enough to > use at run-time (haven't tried this yet... I don't know if the Python > script will just work if it's in the right binary directory). > > I do need to implement some sort of burst/padding special case that I > don't quite understand from the specs yet. :) > > Next to do is DTS conversion. I believe I just need to (1) parse > enough DTS to get the frame size (and then read the frame), and (2) > write out the correct preamble (it's different from AC3). There are > software DVD players out there which already play DTS streams, so I > should be able to take some cues from that. > > The problem with pre-converted files is that they're just WAV/FLAC. > Although they play just fine on the SB2, they won't work in, say, > foobar2000, or (probably) WinAmp. > > I'd like to make a recommendation about how to alleviate confusion with > such files, but I'd like other people's views too. I see two choices: > > 1. Leave the file names as "whatever.ac3.flac" (or > "whatever.dts.flac"). Humans can see the embedded file type, but > WinAmp/fb2k/etc will have problems playing them. SB2 will just work > with a digital output, but won't work with analogue outputs (and it > can't tell the difference between normal FLACs/WAVs and these special > FLACs/WAVs for the analogue output). > > 2. Use a custom extension for converted files, say > "whatever.ac3.spdif-wav" and "whatever.ac3.spdif-flac". (And > "whatever.dts.spdif-wav", etc.) No danger of other software thinking, > because of the file name, that the file is understandable audio. The > 'spdif' and 'spdif-flac' extensions are of course open for debate. > > Question for those in the know: can 'convert.conf' be configured to do > something different depending on whether the player is using digital > outputs? In particular, we'd probably want to convert the audio to > silence for players that can't take it (or possibly get more complex > and find some utility to decode the file and downmix it!). > > Question about SB1: what's the situation with digital pass-through? I > believe there's a firmware bug which causes the data to be corrupted in > some way (bit inversion? Byte swapping?). That would affect these > converted files; I guess it would also affect a standard WAV. Is that > right? A utility to convert the digital files to an SB1-compensating > format would be useful (and not too tricky, I think), but does it need > to be applied to all WAV files or just those with the proposed special > extension? > > I welcome any ideas or answers. SB1 has bit-correct output at S/PDIF output except for a sign reversal of the samples. I get correct passthrough of DTS wav file if the sign of the samples is inverted. Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: digital output format?
seanadams Wrote: > The additional resolution is just ignored by the DAC - it works like a > 16-bit interface. I assume then that the 16 bit DAC uses the 16 most significant bits of the 24 bits transferred by S/PDIF. How does digital volume adjustment work in this scenario? Steinar -- sbjaerum ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
