Posted 9th December 2013 at 08:52 AM byjan.didden Updated 10th December 2013 at 07:02 PM byjan.didden
It’s a recurrent issue: you want to build a preamp, a DAC, a phono stage, anything that needs a nominal supply voltage between 3.3 and 15VDC, positive and/or negative polarity. Sometimes you want several supplies to isolate stages from mutual interference via the power supply. So you want a power supply regulator that approaches an ideal DC voltage source as best as possible within reasonable cost. In your search, you inevitably run into the term ‘superreg’ – so where does the name come from and what is it?
The history of very high performance low-voltage regulators is well documented on Walt Jung’s website (www.waltjung.org – look under Library|Regulators & References). An early design that attracted attention was Mike Sulzer’s, published in 1980 and 1981 in Audio Amateur. I added something to that in 1987, and then I was invited by Walt Jung to work on a further improved version. This was published in a series of four articles in Audio Amateur in 1995 by Walt (part...
Posted 9th December 2013 at 01:00 AM byhbc Updated 14th December 2013 at 06:23 PM byhbc
The EAR 834p, is a marvellous and clever circuit, in its various optimisations and design features that are often not really understood... I acquired my second one a while ago, my first one was tweaked quite successfully to make very acceptable sound by adjusting the 10pf to 5pf and changing output CF to ECC81, and removing some surplus output potential dividers, and careful valve selection. Lowest noise first, and highest gain second.
However, it went away...
The problem I have seen with these is that they sometimes suffer muddy woolly sound in the bass which tends to spoil the over all effect. The new phono box I recently got suffers this way. Now I am suspicious of feedback loops, and how they work. I suspected that the 834p was maybe running a little short of gain in the lower registers. As I enjoy heresy, and after much consideration, I decided that bootstrapping the second stage ought to give me some more gain for the feedback loop to work. See circuit...
Posted 3rd December 2013 at 05:12 AM byabraxalito Updated 18th December 2013 at 02:59 PM byabraxalito
I'd like to be sure that my amp and the passive XOs in my speakers aren't significantly the bottleneck in the sound I'm getting from my DAC prototypes. To get a second reference point I'm starting to build a passive line-level crossover. Here's a pic of it taking shape - its using AD605s because I have plenty to hand and they're cheap and capable of being subjectively transparent with the right power arrangements. The pot core inductors are implementing LR4 XOs in balanced. Using the AD605s with their low input resistance and noise allows manageable inductor sizes for the XO - the largest being 9mH for a 3.5kHz crossover frequency.
Ah - I forgot to mention how I propose to do the transformation from normal (2V) line levels to the 70ohm impedance/20mV paradigm at the AD605 inputs. I shall be winding a pair of ferrite-cored step down transformers with centre-tapped secondaries Lots of turns to look forward to with 0.1mm or perhaps finer wire as I'd like to get at least 5H...
Posted 28th November 2013 at 11:55 PM byabraxalito Updated 12th December 2013 at 07:53 AM byabraxalito
A few weeks ago I built an almost completely passive analog output stage for my DAC/filter front-end and found it sounded noticeably quieter than my active ones which have been using AD605 and AD8129. The only active component in the experimental stage was a single emitter follower buffer run in classA - voltage amplification was achieved by a much higher than usual I/V resistor and passive filter specially designed for the higher impedance.
The comparison left me rather puzzled as to what was clouding the active stages. It turns out it was inadequate power supply decoupling - the fix is an abundance of ceramic capacitors soldered across the top of the chip.
On the AD605, the feedback resistor in low gain mode is a relatively low 820ohms which presents something of a challenge. Also I'm using an inverting stage following with a similar value input resistor. While the low combined values don"t present any significant challenge to the output stage itself (which...
Posted 22nd November 2013 at 12:45 AM byrjm Updated 5th January 2014 at 09:17 AM byrjm(update schematic to 20f4)
Update: I've ordered parts for small number of Sapphire 2.0 kits. The normal price will be $125, but as an introductory offer the first batch will be available for $100. Kit includes a set of boards and all the parts for the board. You need to supply the transformers and diodes, as well as a volume control, and the chassis hardware.
November. That time of year for finally getting around to advancing some of my audio projects a little.
The Sapphire has remained in "rev 1+" for some time now, partly because of time constraints, partly because of the lack of popularity, and partly because it was already a re-spin of the beta version and worked just fine.
There were a few housekeeping things I wanted to add though, which have been included in the 2.0 revision.
- added a dedicated ground (GND) pad to connect to chassis...
Posted 10th November 2013 at 12:00 PM byrjm Updated 10th November 2013 at 02:11 PM byrjm
Last one for today.
I was having problems getting this to work, the key seems to be adjusting R6 to null the voltage offset. Works fine now, but this is just a rough reverse engineer of the diagram, the parts values and the type of transistors are essentially placeholders.
The input jFET is shown as a dual package. The output bias current is most likely much lower than the 100 mA I configured, so the class A output power is proportionately smaller. For the rest of the currents and the types of transistors used, your guess is as good as mine.
edit: R7 should most likely be closer to 47 ohms, while another 5p capacitor should go in parallel with the feedback resistor, R17.
Browsing through an old issue of MJ (No. 1076), I found some sketches of the Marantz SA 11S3 SACD player analog circuitry which seemed interesting.
The are two discrete op amp "modules", one is used for the I/V converter and low pass filter, the other design is applied to both the line output and headphone outputs.
The parts and values are not given, so the circuit below is just a working mock up in LTSpice based on the published diagrams.
The output stage from the headphone / line driver op amp is attached below. There's a JFET input and BJT gain stage as well, and the whole thing is wrapped up in a feedback loop. The buffer itself, being unity gain, works just fine as a standalone circuit element.
Posted 3rd November 2013 at 03:29 AM bygooglyone Updated 3rd November 2013 at 03:31 AM bygooglyone
I noted that the new PIC32 series micro controllers include I2S along with the SPI interface. Well at least a few in the range do. This got me to thinking:
- A 32 bit micro using a fairly efficient RISC architecture
- With I2S in and out
- That runs at 80MHz.
I chose the PIC32MX450F256H.
Surely to god I can do something fun with this. But what?
Ultimately I will try chucking some IIR filters in here to see how they go (there is heaps of processing time available). But in the first instance I want to do a DDS. Reason being that I have more active crossovers than I have speakers (and that is saying something! - ask my long suffering wife!).
One thing that I have been on the look out for is a decent DDS synthesister for audio band that has really low distortion. My current Audio synthesiser uses the AD9952 DDS chip. OK, this runs at 400MHz, but it does use a 14bit DAC, and can be run right down into the...