DANGERDAN

This is the problem i face with the m-dac and i am trying to figure out the possible problems with using low attenuation with this, possible fix would be to use a -20 or -40 attenuation in the line which is a permanent fix which a lot of people have done.

my assumption was you were after a connection to sub woofer in. not too much to gain and more is loos if you do not use the sub woofer in which normally is he THX standard of 80hz. some units have varying xover points, but once you approach 120hz, you loose omnidirectional masking. Thing is i want to be able to cut the mains off at about 80 hz with a slope of about 12-24db as this would be better than just trying to use the crossover on the sub to match the speakers, especially when the slopes on speakers are unpredictable and the frequency response my room gives off with low frequency's due to reflections and resonant distortion, adding a sub without a high pass filter would be troublesome and difficult.

I did ask him about comparison between analogue and digital attenuation and he did go on to say (in his own way) that using the digital volume control at low levels could in duce audible distortion, which i have found people claim but if you used the volume controller at high levels closest to its optimal voltage (2.1v) or 0db on the screen indicator then this should not be enough quantization to be worried about. As i could tell there are benifits to having digital volume control but as you have seen as well there are also cons to this type of topology. Grats on 3500 post's haha. EDIT The digital volume control is often seen as bad because as the volume is decreased, the bits are lost. In general, for every 6 dB attenuation, 1 bit is lost. For example, if the digital volume is set to -12 dB, then 2 bits are lost. This is a serious loss of resolution if the DAC has only 16 bits to begin with. However, this is no longer an issue with the 32-bit ES9018 DAC. With the full 32-bit data path, the digital volume control has more than enough resolution to accommodate the loss of bits. So the loss of 2 bits, as in the example, is practically inconsequential to the 32-bit DAC.

I seemed to have missed your post but it was helpful none the less, i had done a bit of research on the m-dac and spoke to its lead designer about a few aspects over at another forum. He gave me some insight to its technology and concerns i had which was really good, heres a few for those who are interested. I think i will possibly buy the rotel dac to see how it compares in the future. Quote: Originally Posted by DANOFDANGER From what i can read with the M-DAC passive preamp being enabled it is a concern that audio quality may be attenuated at low listening levels, could you state why this is? John: The MDAC does not have a passive pre-amplifier - its a Digital pre-amplifier so the audio signal is scaled in the digital domain - there is no change in the Analogue gain structure. Having too much system gain requiring you to listen at high attenuation levels is a bad thing. "Gain" is never for free, each gain stage causes a reduction in sound quality - there's no such thing as a 100% perfect gain stage. When I hear of systems where the owners are listening with level settings of -60dB or lower this suggests to me a very badly matched system. Quote: Originally Posted by DANOFDANGER I would have thought since the volumes were digitally modified directly from the DAC that it would inhibit no audible distortion. John:Simple question is how to "Reduce" something without loosing anything - you will always be reducing a signal into a "fixed" noise floor level. You have a large bucket of water, and want to pour it into a smaller bucket - the smaller bucket will eventually overflow, and you loose the rest.... Bigger into smaller does not go. Quote: Originally Posted by DANOFDANGER Or does it have something to do with the analog end like at the operational amplifier or other analog components. Is this because the DAC was designed optimally at a certain voltage level ?? John:Both optimised Analogue and Digital systems (once converted back into the analogue domain) will always face the same noise floor issue - 24bits audio is BELOW the theoretical noise floor caused by the random movement of electrons "heat" energy, unless you live near absolute zero... Digital domain attenuation when done correctly should be no different to performing analogue domain attenuation - however practical "real world" implemention issues need to be considered. With Digital attenuation, the Analogue system is always operating at Full gain - turn up the volume knob of any analogue amplifier with no music playing and you can hear a slight background Hum, Hiss, RF intermodulation products etc. though the speakers - this is a reality of analogue electronics. Any Digital product by there very nature will produce RF energy. There is a practical limitation on how much you can filter this energy before you start to detrimentally impact the audio quality. Without any form of analogue attenuation this "leakage" RF energy from the DAC is pumped directly into the Amplifiers input stage which can then be demodulated into the audio range. Transistor inputs stages are by far the most sensitive to RF demodulation by a significant margin followed by Jfets, Tubes then MOSFET's. Adding Analogue attenuation in front of the amplifier reduce both Audio AND RF energy - this does not happen with pure digital attenuation. Quote: Originally Posted by DANOFDANGER Another thing if you disabled the passive preamp stage and say connected it with a windows computer and controlled the digital volume from there, would this have the same adverse effects of sound degradation or would this help with the distortion at low levels. John:NEVER use windows level control - the algorithms used by the Windows sound kernel to achieve digital domain attenuation are truly horrid - unfortunately, Windows digital domain attenuator is a poster child on how digital attenuation should NOT be done. The MDAC also has the advantage that the attenuation can be performed in a 32bit domain (in fact greater then 32 bits). John Originally Posted by DANOFDANGER Hey thanks john that really helped me, one last thing as i am about to go to the atore and buy one now.. so i should leave windows volume control to full so theres no attenuation distortion coming from windows bad algerithm. John:Yes - thankfully setting Windows level control slider to 100% can result in Bit accurate data if everything else is configured correctly John

Na i knew that but i was wondering more about the crossovers, being able to configure the crossover properly but without a filter from a receiver of some kind to filter the mains i don't see it possible. I could just use the LPF on the subwoofer to match the rolloff of the mains but wouldn't it be better cut off the mains higher with a HPF so that the sub does most of the lower frequency. ??.

iv been doing some thinking and how in the world are you suppose to configure a sub with dac units, you are only left with the subs integrated lowpass filter but nothing for the mains ?????.

ordered a audiolab m-dac which should be here in a few days, from what i have read its a really nice unit. so we shall see how it goes.

Iv never heard of studios EQing their sources (at the final stage), maybe the equipment but i would guess it would be to again flatten out the tonal balance distortion set by their equipment. With regards to horn directivity i believe your talking about a different type of EQ are you not ?, in the actual design of the horn where the CD design has trouble with roll off and also sensitivity so in the crossover they implement a attenuation and eq filter ??. At least i would think they already have a filter in our klipsch so CD EQ would be pointless no ?. I still hold true that EQ works best with source correction, You cannot completely correct a bad speaker as well as a bad signal and sure it can improve it but its more like masking tape as i said before and you would DEFINITELY get better results with a speaker upgrade.

I tend to disagree, EQ's is more like masking tape in a job where nails are needed. They change the characteristics of the sound but not the formality or structure, by that i mean EQ's can only resolve so much before they add their own distortion to the sound and you reach a limit of configuration. Studios and master recording buildings don't use EQ's on the final output stage of what they are working on, they have individual recordings of which they can manipulate without affecting other instruments. If your dealing with room acoustics that change the sound you would be best off treating the room than using EQ's as the former would have a much larger and more linear outcome than trying to just adjust the signal and if your speaker is the source of bad frequency response then you can only so much try to improve that with a EQ in which if you want overall better response you would upgrade or change your current speakers. Where EQ probably works the best is when the source adds coloration that you don't like but in terms of accuracy i don't think the EQ has that much of a impact on it, so IMO if your ever going to use a EQ its to change the tonal balance but never the accuracy of the frequency response. Smileys all round :)

When you look at the way most dacs work, the pre-ringing (which most of them have a significant amount of) is where many filter makers are trying to act. The DacMagic allows you to change the filter type (including minimum / linear phase) and one of the filters alows you to remove a fair amount of the pre-ringing. Several digital components are using features such as apodization (Meridian for one). There is some sense behind doing this and as with speakers, you have to decide which really sounds right since none of the digitial recreation is considered to sound right. How many people use loudness or bass and treble controls or even more heavy handed, audyssey? Yea i understand and i even use EQ's to balance out incorrections but i still believe you shouldn't have to, Id like to think if you had the equipment that was linear enough EQ's or filters would become obsolete. Also it strays me from playing with the dam thing all the time as well.

Iv got my decision down to three dacs, all of which are available to me as most others are not. DACmagic plus Rotel RDD-06 Audiolab M-DAC The dacmagic seems to be the least of my favorite due to its layout of trying to upsample (forcefully) and i just don't agree with their agenda, that and having filters available to the user seems to me they are trying to manipulate the signal more than actually trying to reproduce something more linear. The real decision is between the Rotel and audiolab, both are beautiful and support many specs that i favour in a unit like this, the Rotel for example hosting the special and highly regarded God of all DACS which is the wolfson 8741 but doesn't have a headphone support/amp. The Audiolab however does and has another chip i see favoured by people in the audiophile scene, its a little more pricey but that's because of the headphone/preamp stage added to the circuitry. So out of the two really as i am not too keen on the DACmagic i have a big decision to make and a bit more research to do as well.

Can anyone tell me if the Dacmagic plus has a passive preamp or not ?, it states its a digital volume control so is it classed pre amp at all due to actives usually are potentiometers with pre amplification stage and passive just being a analog variable resistor (pot). So .....

Thanks, yea i expect as much due to my old marantz barely pushing 20 watts, only was concerned about the new circuitry design difference. XD

Definitely many to choose from, thats why its good to get ideas, a lot of these are not as widespread as others and difficult to find. Thanks.

From what i can see they supported 16 bit @ 48khz ?, whats more to need than that. Any higher and you gain nothing its just myths.

Thanks for your input.

From my research and discussion i never found anyone who could prove that jitter really made a difference, i think that the distortion that people claim jitter has on a signal is still far below any snr distortion left over by the chain so its in audible. That DAC you showed looks top notch il have a look into that thanks.

the computer can use USB which is superior to optical or coax. um it should all be the same, if anything usb would be the least qualifying due to it having the least capable bandwidth overall.

iv seen a bel canto around before and was interested, i might look for one. what model of the grant fidelity did you mean?

I understand about other contributing factors such as circuitry design, op-amps quality capacitors etc but the signal is only as good as its weakest link, so a good dac chip is needed in order for the rest of its circuitry to be beneficial wouldn't you agree ?. I am wanting to spend about 700-1000 US on a nice dac. EDIT forgot about interconnects, i need two optical connections, one from the tv and one from my computer.

Hey there, being a believer in Quality DAC's i am looking for peoples opinions on DAC's they have tried. I have had my fair use with burr browns multiple and common chips around today, here are a couple. Burr-Brown PCM1796 (D2x) Burr-Brown PCM 1792A (STX) Couple of head units with burr brown (JVC,ALPINE) I am looking atm into a DAC magic which has a wolfson 8740 and i have not listened to any wolfson before but i hear great things so i am excited, i also see a lot of great reports of cirrus. Anyone know any they have heard of before or know any with perhaps some of the high end dacs, i would like to find some with a wolfson 8741 as they are praised as flagship. xD

Yea i guess the only way to know is to fit a power meter on and find out.

Thanks for your reply, this was similar to what i was thinking however i was under the impression that when no signal is present that the class B state has the possibility to switch off the transistors and when a signal is fed the class A state is active on low levels but class B is introduced at higher levels of power output. The heat seems to correspond to that as well with the heat being more present at higher sound output levels and at lower levels there is less.

Considering the design of class A and its inneffeciency i am wondering how much power the unit does when its in standby with no signal on, i feel a bit of heat coming from the unit and thought that with class A/B architecture that it would use no power. Also with Klipsch being so efficient wouldn't the amp be using a small amount of power. Thanks.

So a alternative technology to variable resistors/potentiometers ?