FYI, For those with M-audio delta 44/66, I found an interesting article: http://www.sm5bsz.com/linarch/linarch02-03/msg00313.html which I uudecoded and placed here: http://nielsmayer.com/improving_noise_floor_delta_a2d_conv.doc "Improving the noise floor of a Delta44 A/D converter (Sept 3 2002)" Any comments or experience with this mod?? ............................. Regarding the previous thread, the newly won, potential victim of the quasibalanced output mod: "Terratec DMX 6fire 24/96" http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=180514801725 The attempt: silencing the noise out of my nec xr385/yam db50xg ( http://www.megatrade.ru/English/db50faq.html ) which i think is the result of lack of shielding, bad regulated power supply, or crap dynex dx-sc51 that has a waveblaster port, but leaves the db50xg card (about twice the size of the new-tech soundcard) hanging in the middle of your computer, without even the groundplane of original ISA-full-length card to protect it... ...... it's basically switching/ground/digital noise unless someone has a better explanation for sub-50hz noise in a country with 60hz line voltage: http://nielsmayer.com/db60xg-jaaa-L-noise.png shows the noise level at -51db whereas the right channel is the more-reasonable -65db http://nielsmayer.com/db60xg-jaaa-R-noise.png http://nielsmayer.com/db60xg-japa-L-R-noise.png (note the db50xg itself claims "18 bit output D/A's" so the noise, in theory, should be well below -96db) ....... The hope: the new Terratec DMX 6fire 24/96" houses the db50xg in a separate breakout box, and has halfway decent A/D section, and probably slightly less sucky input and output analog components than the equivalent M-audio delta 66 (both cards are supported by Module snd-ice1712), and probably many times better than the cheap dynex dx-sc51 (Module snd-ice1724) currently hosting the synth. https://vsr.informatik.tu-chemnitz.de/twiki/bin/view/Sonstiges/WaveTableSou… "Terratec DMX 6fire 24/96 PCI inside the breakout box, Good as of 2005!" http://ftp.terratec.de/Documentation/Linux_and_Mac_OSX_Compatiblity.pdf "the 6fire uses the ICE Envy24, which is supported under Linux by the Alsa project. The gnu application envy24control offers a very good control panel derivative. http://www.soundonsound.com/sos/Apr02/articles/terratecdmx6.asp http://www.hardwareheaven.com/other-tech-news/640-terratec-dmx-6fire-review… http://www.soundonsound.com/sos/mar05/articles/qa0305_6.htm Niels http://nielsmayer.com

Fons: Thanks for your suggestions and input. Here is are some more outputs from a different dynex dx-sc51 soundcard: http://nielsmayer.com/npm/jaaa-db50xg-L-fullband-captureoff.png http://nielsmayer.com/npm/jaaa-db50xg-L-fullband-captureon.png http://nielsmayer.com/npm/jaaa-db50xg-L-lowfreq-captureoff.png http://nielsmayer.com/npm/jaaa-db50xg-L-lowfreq-captureon.png http://nielsmayer.com/npm/jaaa-db50xg-R-fullband-captureoff.png (note interesting anomaly at ~14K) http://nielsmayer.com/npm/jaaa-db50xg-R-fullband-captureon.png (anomaly disappears in the noise) http://nielsmayer.com/npm/jaaa-db60xg-R-lowfreq-captureoff.png http://nielsmayer.com/npm/jaaa-db50xg-R-lowfreq-captureon.png Capture on vs capture off on the soundcard shows a big difference in the noise floor. Is the -62db/hz&-66db/hz at 0.1hz an anomaly? In 'jaaa' the 0.1 hz value "jumps around" as if there were a shifting DC level. It can be seen in 'yass' for example... "capture off" shows a flat line with AGC turned on. "capture on" shows a noise that is entirely positive voltage, which can be raised higher by adjusting the capture gain above 0db. Thus the purchase of a used $20.00 soundcard to replace this $0.99 one. You gets whatcha pay for... Niels http://nielsmayer.com

Last time I saw a random peak that strong, it turned out to be a byproduct from an opamp that was underdamped and self-oscillating at about 10MHz... It moved around just a little bit as the oscillation wasn't itself entirely stable. Complete shot in the dark, I do't expect to ever see it again (except in homebrew designs that are slapped together without alot of design analysis). Ah, the fun of using sound cards as analyzers :-) Monty

FYI, I just got and installed the Terratec DMX6Fire -- definitely a better alternative for Linux as you can use "envy24control" to control the card, instead of the confusing and error-inducing alsamixer. Plus you get an extra MIDI I/O, spdif and toslink I/O, etc. Overall, an excellent $20.00 card, perhaps even preferable to the equivalent M-Audio Delta 66. The card's performance seems in line with it's specs ( http://www.soundonsound.com/sos/Apr02/articles/terratecdmx6.asp ) "A-D converters: 24-bit 64x oversampling (part of AK4524 chip). Input dynamic range: 100dB (converters only)." With the waveblaster connected to the db50xg, but no output coming from the synth-card, here's what the noise looks like: http://nielsmayer.com/npm/jaaa-DMX6Fire-L-fullband-db50xg.png http://nielsmayer.com/npm/jaaa-DMX6Fire-L-lowfreq-db50xg.png http://nielsmayer.com/npm/jaaa-DMX6Fire-R-fullband-db50xg.png http://nielsmayer.com/npm/jaaa-DMX6Fire-R-lowfreq-db50xg.png (above tests set with input amplification for the A/D set at 126/163 in envy24control, which seems to handle the full-output of the loudest of my XG-midi test files without clipping... you can make the noise disappear entirely (or at least go below -160db) by turning the A/D amplification below 60/163.) Subjectively, however, the sound is not better! Although cleaner and more noise free (the other dx-sc51 would light up the lowest level -60db output metering LED's on my external DAC ... this one doesn't) this card doesn't actually sound as "punchy" as the same synth playing through the crap Dynex dx-sc51. Especially interesting is that the card sounded particularly anemic&shrill when I first plugged it in, but as it ran overnight it now sounds much better (electrolytic capacitors might have been in storage for a while on this used card). (For all my listening tests, I'm using TOSLINK outputs into a higher quality external D/A, so the only change is the sound of the A/D and analog inputs, not the D/A or outputs). By lacking "punch" i mean that the bass-transients sound somewhat "round" and "soft" . Bass drum doesn't have a forceful "kick" to it. And likewise a there's little "snap" to the attack portion of the bass. On the other hand, the high-end sounds a little brighter and less "veiled" but sometimes shrill. To improve the bass-response, I'm considering changing some of the input decoupling electrolytics to larger values. The JRC4580 op-amps are surface mounted and therefore difficult to replace. (Nonetheless, people have replaced the opamps and caps on this board and claim/measure an improvement: http://forum.rightmark.org/topic.cgi?id=4:122 ) Perhaps the input op-amps on the Dmx6fire are "slew limiting" or rounding large-signal transients. The cheap Dynex card didn't have *any* input opamps and connects inputs directly to the Codec via a coupling capacitor and resistor. The input opamps on the Dmx6fire, on the other hand, allow for adjusting input levels. On the dx-sc51 the input level could not be adjusted and would easily clip if the main-volume control of the synth-card wasn't turned down. Of course, with the external breakout box on the dmx6fire, it'll be easier to perform a simple mod and get back to hearing the excellent sound of this synth -- bypass the soundcard entirely (other than using it to house and midi-drive the db50xg) and solder 1/4" TRS jacks driven by a RDL STA-1 active transformer pair.... But first, I'm going to let this card "burn in" for a week (even though it's 7 years old) as that seems to be improving the sound without my doing any work, or spending any money :-). -- Niels http://nielsmayer.com

On Wed, Jun 9, 2010 at 8:37 PM, <fons(a)kokkinizita.net> wrote: It can be surprising how often this is the case -- differences that you can describe, articulate, and explain in normal listening can surprisingly often turn out not to be distinguishable at all under proper test conditions. Embarrassingly so, sometimes. You should try to set it up (Niels), not so much because I think you're wrong in this case as because you seem open to testing things properly and this adds a rather compelling dimension. Smiley apart, this is also not entirely unreasonable. Every time I listen to a vinyl record I think how enjoyable lower-quality sound[*] can be. Didn't some recent study show that kids these days prefer mp3s to raw audio? Back in the day I used to like the squishy sound of tape cassettes, and they still have a certain something -- particularly for bigly-produced 80s metal. Mmm. *digs out old Scorpions tapes* Chris [*] if you find anyone who still believes vinyl is higher fidelity, try recording their favourite record to a digital file with a decent soundcard at a sensible sample rate and bit depth -- it should be completely indistinguishable from the original record. I've tried this and convinced myself at least. Doesn't stop me from playing and loving records, though!

On Wed, Jun 9, 2010 at 10:03 PM, <fons(a)kokkinizita.net> wrote: Purely subjectively, that whatever it is that constitutes "the magic of the vinyl sound" is still there when recorded onto a computer, and indeed even when encoded to mp3. (assuming decent hardware and quality settings throughout.) I haven't actually proven that I can't distinguish between a record and a digital recording of that record... just done enough to silence that bit of the vinyl mystic in me. Even so, I still like "the act of putting on the record". Chris

On Wed, Jun 9, 2010 at 12:37 PM, <fons(a)kokkinizita.net> wrote: I actually checked that peak at 116hz and figured it couldn't be the rectified result of the US-grid 60hz AC. That means we'd be running at 58Hz. ?? Since that's unlikely, what's the likelyhood of a 2hz frequency error in jaaa?? 127 causes clipping. [[side-annoyance: why do some cards clearly light up the red "overload" LED on the external DAC, and this Terratec DMXFire doesn't? I end up using 'qarecord --jack' as a meter w/ a clear red overload indicator that i can actually see, but I wish there was a simple Qt notification-area gadget that would just turn red when you overloaded an input or hit 0db]]. Indeed. The card has stickers on it indicating it was produced in 2003. How old is too old for a decoupling electrolytic?? Have any suggestions in place of the no-longer available http://search.digikey.com/scripts/DkSearch/dksus.dll?vendor=0&keywords=… ?? And having to resort to something like this seems pretty wanky for a $20.00 soundcard: http://www.tnt-audio.com/accessories/auricap_e.html It's not really an issue of "bad design" it's an issue of compromise at a given price point. How much bandwidth does the op-amp have, and how much negative feedback is being applied? But a wide-bandwidth op-amp is going to be intrinsically noisier.... and a quiet op-amp is going to be intrinsically slower, and improving either makes them expensive. Actually, one of the improvements people get out of replacing op-amps is IMHO the output slew-rate and overall bandwidth spec. Thus the stock JRC4580 opamps on the Audiotrak Prodigy HD2 are socketed, and upgradeable: http://www.head-fi.org/forum/thread/322497/audiotrak-prodigy-hd2-opamp-conf… has a lot of mention of bass "punch" and "control" improvements from replacing the op-amps as well as making the highs less shrill. If you really want to go overboard, they have discrete replacements for these op-amps: http://www.itemaudio.co.uk/prodigy_hd2.html http://bursonaudio.com/burson_opamp.htm Here's the "before" : http://nielsmayer.com/DjColtraneRexx-qchord-salsa.flac (sorry, there's some overs, this is before i learned of the dynex dx-sc51's easily overloaded waveblaster input, combined with potential for large outputs from the Yamaha db50xg (actually NEC XR385)). At some point I'll have both soundcards installed in two difft computers and two different NEC XR385 synths (got a second one for my son coming over from china for $13.00 + 5.50 shipping) and I'll have two files to listen to incase you want to hear what "punch" versus "lack of punch" sounds like. I don't have time right now to go find the section of MIDI I used earlier and re-render using the new soundcard.... eventually I'll post a followup to this thread w/ more results. I agree that when it comes to measuring electrical signals (as opposed to human perception of sound, or any human interpretation of the world around them) there is no place for handwaving, mysticism, etc. Unfortunately, the science of perception is wholly unfounded and mostly bogus if not outright scientific fraud, thus leaving plenty of room for every kind of interpretation. [[ please read and understand this book before having too much of a knee-jerk reaction to my statement http://books.google.com/books?id=2sRC8vcDYNEC&printsec=frontcover&s… // http://books.google.com/books?id=2sRC8vcDYNEC&dq=understanding+computer… ]] However in this case, my hypothesis is simple. No handwaving, hermeneutics or mysticism is needed. Coupling capacitors induce a low frequency phase shift that is usually ignored because "engineers" incorrectly care only about real-plane/power information; it is misguided to think that humans cannot perceive phase information and "imaginary plane" information. More simply put, in reproducing a picked electric bass, you expect the the "pick noise" (high frequency) to arrive at the exact same time as the the onset of the low-frequency fundamental of the note. The coupling capacitors delay this onset. The simultaneous overlay of fundamental "attack" and "high frequency pick/plucking" onset is what gives a bass it's "snap" and "pop". If you delay the attack of the low-frequency fundamental, you get two sequential softer attacks, and you get a bass sound that is "muffled" "round" or "anemic"... A feeble analog output circuit might not be able to dump enough current out of , or into, the coupling capacitor to allow it to quickly transition from the lowest to highest voltages within the output swing. And there may be stray inductances on the circuitboard or in the capacitor itself that induce oscillations. In which case the output impedance of the op-amp needs to be linear at both low and high frequencies in order to damp out oscillations. Those are precisely the characteristics lacking in low-cost op-amps. So, unless gain is needed, perhaps the cheaper soundcard gets the best sound since it has no op-amps, no bandwidth limitations crossed with negative feedback, no output slew rate limitations, etc. Just a capacitor and resistor in front of an 18 bit ADC. Since a single device inches away is driving the Codec, there really is no need for the op-amp electrically -- there's already an output op-amp and coupling cap on the db50xg (probably a JRC4580) after its 18bit DAC. More interestingly, since I'd rather get away from the subjective and back towards the measurable, what kind of tests would be an appropriate predictor for the "bass punch" of a soundcard? Certainly not the frequency response or noise characteristics. More likely it would involve something like reproducing a low-frequency full-voltage squarewave, and observing the slew rate and ringing response caused by bandwidth limitations in the op-amp combined with it's own negative feedback.... -- Niels http://nielsmayer.com

On Wed, Jun 9, 2010 at 3:05 PM, <fons(a)kokkinizita.net> wrote: Thus my empirically derived value of 126, which is nice to know is a tiny bit below unity gain. Also, FYI, the gain issue may be ultimately what's causing the difference between soundcards. The Dynex dx-sc51 was actually easily overloaded by the standard amplitude output of many of the XG-midi files available on the net, which means in order to hear the db50xg without distortion, I had to turn down the master volume on the synth -- there is no other gain reduction available on the input. However, the Terratec DMX6fire24/96 doesn't have this problem, plus it does allow for gain reduction. Therefore, with the terratec, bigger voltage swings are happening. The S/N ratio is improved because the signal is bigger. In other words If I really want to fairly A/B test the two cards, I need to set the volume output by the db50xg to the exact same low level as needed for the dynex. and them boost the gain on the input of the soundcard to compensate back to 0db full scale swing. After all, a full-swing signal would be testing the slew-rate limitation of the synth's own output opamps and caps, in series with the slew-rate limits on the card's input opamps. And also, I truly think burn-in and not my ears adjusting is the reason why this card and synth now "thump" like it's supposed to, after playing full-volume through the soundcard for 20+ hours nonstop. And you can't just eq away a lack of thump (i tried): just makes the bass louder and more "muddy". (although you could go all "aural exciter" ( http://en.wikipedia.org/wiki/Exciter_(effect)#Aural_Exciter ) and try delaying the high-frequencies to match the phase-delayed bass) I think this would be an excellent area to investigate with your tools. Almost a better use of the scientific method than whole soundcard testing because you can focus on a single variable that is known to be problematic. And unfortunately, electrolytic capacitors are the one component that are about as far from their theoretical ideal of any component. In fact, i think it's totally braindamaged to be using electrolytic capacitor for audio coupling... but then again, a non-electrolytic coupling capacitor would be huge, and would also probably have a ton of self-inductance given the amount of conductor needed to make a large value capacitor using a standard dielectric. IMHO, the next advance in circuits would be totally DC-coupled designs, with built-in microcontrollers to auto-trim away any DC offset.... they should just integrate the whole thing into a new "super op amp" containing a digital control "watchdog" that automatically adjusts its DC offset, output impedance, output damping factor, etc... I don't dispute this, however, can meaningful impulse responses be made of an entire soundcard from analog input->A/D->DSP/Mixing->D/A->analog output? It's giving me a headache thinking about even a simplified case -- an analog sample-and-hold circuit. Isn't "sampling" anti-causal? Yes, such circuits are easy to understand and model: http://en.wikipedia.org/wiki/Low-pass_filter#Passive_electronic_realization IMHO, if done correctly, the significant phase and LF rolloff will happen entirely outside of the audible spectrum. So it's really a matter of calculating the input resistance, and coming up with a capacitor value large enough that the rolloff is below 5hz or so .... Which is probably how http://www.jensen-transformers.com/an/an003.pdf came up with the 220μF value for the coupling capacitors, versus 47μF typically used. -- Niels. http://nielsmayer.com

On Wed, Jun 9, 2010 at 3:05 PM, <fons(a)kokkinizita.net> wrote: Actually there is a relationship, and it is certainly not simple. However the generality holds, mostly because when you get down to it a fast device achieves speed through either size, or high-current. High-current induces one kind of noise, and small device geometries induce another; field-effect, it's own, alongside material device physics, temperatures, etc... For example one might prefer FET or MosFet for linearity and gain-bandwidth linearity, but they're noisy. Ultimately it comes down to gain-bandwidth, which is nonlinear in low-noise Bipolar Junction-based opamps like you'd normally use in audio: http://waltjung.org/PDFs/WTnT_Op_Amp_Audio_3.pdf http://focus.ti.com.cn/cn/lit/an/sloa051a/sloa051a.pdf "Selecting High-Speed Operational Amplifiers Made Easy" ........... Although no op amp is ideal, modern processing techniques yield devices that come close, at least in some parameters. This is by design. In fact, different op amps are optimized to be close to ideal for some parameters, while other parameters for the same op amp may be quite ordinary. It is a trade-off. Some parameters can be improved, but only at the expense of others. It is the designer’s function to select the op amp that is closest to ideal in ways that matter to the application, and to know which parameters can be discounted or ignored. [...] Op amp design is a series of trade-offs and an improvement in one parameter is always accompanied by degradation of others. This is certainly the case with current-feedback amplifiers where, to a degree, dc performance has been sacrificed to gain speed. ............. -- Niels http://nielsmayer.com

While cluing-myself before I commit soldery on working electronics, I finally figured out what is going on with the Dynex dx-sc51 from earlier in this thread.... I was investigating whether I could perform my I2S-vampirism experiment on the soundcard least likely to cause sadness and loss at it's untimely departure -- my Dynex dx-sc51.... (see http://old.nabble.com/digital-upgrade-of-older-equipment-via-I2S-with-jitte… ). Before I butcher the card any further -- which would be to add a stereo balanced out to the db50xg DAC that I've already tapped into -- I wanted to document where I'm at, and post some links to MIDI files rendered using the NEC XR385 (Yamaha db50xg/MU-10 clone) in a Terratec DMX6Fire -- from 2003 -- versus a Dynex dx-sc51 (from 2009) with worse noise specs, most of which appears to be DC-offset. However, please note, just with the trivial analog modification, there's a dramatic improvement in sound, with the analog-modded synth-card in the cheaper "consumer" dynex ($0.99 on ebay) sounding better than a "prosumer" Terratec DMX6Fire 24/96 ($20.00 on ebay). Noise be damned, because it's all DC-offset. Jnoisemeter says the terratec+yamaha have a -75db noise floor regardless of "DC" filter. At about the same estimated amplification level (+12db) the dynex+yamaha have -74.5db noise floor as long as DC filtering is on. But -34.5db without DC filtering and otherwise "flat" response (!). Despite the DC offset (which was also present, more asymmetrically between channels, before the modification), the sound improvement comes from getting rid of unnecessary analog buffering and decoupling -- I've bypassed THREE 10uF series capacitors, a BJT op-amp, numerous series resistors, several 0.1uF bypass capacitors -- no need since there's a known load and short connection needed between the DAC and the Codec of the soundcard. Replaced it all with a single 100uF capacitor. Now the bass thumps properly again, and the fundamental tone is back and back without a "rounded" attack. As bonus, removing the synth-card's output buffer op-amp solves the problem of outputting too much volume and making the sound less distinct and clear. Now I can make useful gain adjustments on the dynex's capture-gain in alsamixer -- the normal +9-+12db of gain to get the digital levels up to 0db is right in the middle of the 0-22.50db of analog amplification provided by the codec. Before I had to turn down the MIDI volume to prevent clipping. Now I can adjust the capture gain upwards of +9db if I want to see clipping on the loudest of publicly available MIDI files. So ultimately, the modification allows you to make optimum use of the dynamic range of the 16bit/44.1K digital source of the synth. The Dynex's analog section is predicated on something I didn't expect to find with a "modern" soundchip like the Via ICE VT1723 -- "Via ICE VT1616 6-Channel AC97 Codec with S/PDIF." -- AC97 means that the analog section is running at 48K, no matter what. The strange low-frequency artifacts presented in the .png's I posted earlier were perhaps the result of resampling from 44.1->48K (?), and a huge DC-offset that is either the result of a bug in alsa-driver-1.0.23/alsa-kernel/pci/ice1712/ice1712.c (used for snd-ice1724) or a leaky capacitor, or the need for a shunt resistor to ground (next mod, along with using -- http://www.rdlnet.com/pdf/Data_Sheets/sta-1.pdf -- sent into "floating" balanced output.). With intention to improve the sound, I completely bypass the NEC XR385's analog output section that decouples the DAC output with an op-amp ( http://www.datasheetcatalog.org/datasheet/nec/UPC4570.pdf ) delivering unnecessary gain, distortion, and clipping-potential to the synth's output. That is, the analog outputs of the AK4510 ( http://electro-music.com/forum/phpbb-files/ak4510_codec_683.pdf ) are connected to the cut-off, 8cm end of a shielded 4-pin analog-cd cable normally found in consumer pc's to connect the cd-rom sound -- bypassing the 10uF capacitors on both input and output side of the synth-card's uPC4570 opamp. On the Dynex card, I plug the 4-pin connector into the 4-pin analog-cd input. In the original design, the dynex's 10uF bypass caps fed into the codec; I replaced with the freshest 100uF 16v capacitor I had laying around that would fit. There was also an RC filter composed of microscopic surface-mounted 330 ohm resistor and 0.1uF to ground that I removed off the CD input. In summary, the synth's AK4510 outputs straight through 8cm shielded cable, to 100uF cap, and on to VT1616... . I left the 10uF ground coupling capacitor and it's 0.1uF bypass between the synth-card and the analog cd in. The original files were recorded at 44.1k: per spec-sheet AK4510 codec uses "11.2896Mhz" MCLK (@256fs), for an fs=44.1Khz, SCLK=1.4112Mhz (@32fs)" , where 11.2896Mhz * 3 = 33.8688Mhz: the crystal near the main Yamaha VLSI is == 33.868Mhz., meaning the Yamaha's sample rate is 44.1K. Recording at 48K, as was done on all files below named "*on-analog-modded-dynex*", is thus a waste of space. It was done to bypass any resampling that would happen by running the codec other than the prescribed AC97 48k.... so it's not a fair comparison, but also, I don't think a higher bit-rate is all that happened to the sound on the dynex -- it's also a combined reduction of all the op-amps and capacitors in the signal path. All files were encoded with "oggenc -q 10": http://nielsmayer.com/npm/Intelligent-Summer-on-analog-modded-dynex+12db.ogg http://nielsmayer.com/npm/Intelligent_Summer.ogg (orig http://www.megatrade.ru/Midi/Isummer.zip -> Intelligent Summer.mid) http://nielsmayer.com/npm/Is_god_-_Free_demo_Copyright_Yamaha_-on-analog-mo… http://nielsmayer.com/npm/Is_god_-_Free_demo_Copyright_Yamaha_-_XG.ogg (orig http://www.blueman.name/Jazz.php -> Is god - Free demo Copyright Yamaha - XG.mid NOTE: demo of the guitar sound capabilities extolled in http://en.wikipedia.org/wiki/Yamaha_XG ) http://nielsmayer.com/npm/Skyline_Herbie_Hancock_on_analog-modded-dynex+13.… http://nielsmayer.com/npm/Skyline_Herbie_Hancock_TG300B.ogg (orig from http://www.blueman.name/Jazz.php -> "Skyline\ -\ Herbie\ Hancock\ -\ TG300B.mid" ) http://nielsmayer.com/npm/chpn_op53-on-analog-modded-dynex+13.50db.ogg http://nielsmayer.com/npm/chpn_op53_coggie.ogg (orig http://www.piano-midi.de/midis/chopin/chpn_op53.mid ) -- Niels http://nielsmayer.com PS: TODO: investigate whether alsa-driver-1.0.23/alsa-kernel/pci/ice1712/ice1712.c has a bug in that the following feature of the VT1616 codec ( http://www.datasheetarchive.com/pdf-datasheets/Datasheets-3/DSA-48587.pdf ) is being ignored: