INITIAL POWER MODS
The power on/off POP seems benign if listening through headphones. The power-on pop is not loud at all. The power off pop is loud. If connected directly to an amplifier, it can potentially damage the loud speakers.
During Power off, the positive rail drops faster than the negative rail and this creates a voltage imbalance resulting in a huge POP. At some point, the device is practically powered by the negative rail [link]. The effect can also be visually seen:
The left LED is on the positive supply. It goes dim first when cutting the external power.
I attempted to balance the power consumption for the two rails hoping to reduce the turn-off POP.
Adding capacitors to the positive rail
By trial and error, incrementally adding capacitors up to 13,000 uF I was able to balance the power decay between the positive rail and the negative rail by visually matching the turn-off rate of the two LEDs in the supply. The POP was somewhat reduced (using headphones) but not eliminated.
Adding resistors to the negative rail
I then tried to match the current consumption of the negative rail with the positive rail. The power consumption of the DAC is: 
- Positive Rail: .18A, 10V
- Negative Rail: 0.06A, 10V
I added enough resistors to increase current consumption by 120 mA. I used 10 1 Kohm resistors on the -12V leg of the supply. Each would dissipate 12 mA of current and .144 W of power -they are rated at .25 W, so we are safe.
The result was better than adding capacitors to the positive rail, the POP at power off was further reduced but still not eliminated. I liked this mod better than adding capacitor, so I made it permanent in the supply.
Note: only tried with headphones. And yes it is a waste of energy, but it is only 1.4W :-)
BYPASS THE BRIDGE?
An apology to my readers for putting up wrong information.
I took a closer look at the J2 connections on the backside and the board and the +/- analog power connections are connected to the power lines after the RC filter. If you power through J2, there will be a 12 ohm resistor to the filter/smoothing caps. This would not destroy the board, but it is not the right design either. Thus there is no good reason to power through J2.
While examining the back of the board and realizing that J2 cannot be used as input power, I figure it is the perfect place to add capacitors to the power lines. Decided to add 330 uF Oscons to the digital 1.2V and 3.3V and Panasonic FMs to the +/- analog lines. This mod is completely reversible. Just cut the leads. I first soldered a row of pins and the capacitors are soldered to the pins.
Adding capacitors to the analog lines further improves the filtering because it is an RC filter.
There is still the +/- 5V lines but there is no more space. I chose not to add capacitors these lines because the 5V regulators already have large output capacitors, and because I thought they would benefit the least: if you examine the backside of the board, you’ll notice that the connectors at J2 trace from different places in the board. The longest traces are from the +/- 5V regulators.
These power mods are “approved”: [link]
If you insist to improve the on-board power supply, try replacing the 6 electrolytic capacitors with aluminum polymer types, 1000u 16V exist in same 10mm SMD footprint and t.ex. digikey stock them at $2.20 each. Should be easy to replace.
You can also add a small polymer electrolytic on the 3.3V output, but please note that the clock oscillator power already have a filter, so I doubt it will make any difference.
Didn’t want to remove the existing capacitors and risk damaging something. It is too early to do any kind of surgery on the board :-). Plus the existing filter capacitors don’t seem quite easy to remove given that there is very little space between them.
FULL DIGITAL INPUT ISOLATION
These are the Silicon Labs digital isolators on the board [link]. I believe the original ones in the engineering board were TI isolators [link]. Both isolators use the same capacitive coupling technology. I suspect the Si parts have equivalent performance when it comes to noise isolation.
According to the diagram below, the parts can also be used for logic level translation and thus they can be powered with 5V on the input. This will make the input compatible and tolerant to 5V signals such as those present in a serial interface of a standard Arduino microcontroler.
5V tolerance is also needed if interfacing to the I2S/serial signals of older generation CD players when they used spend good engineering money in making CD-only players. I like older CD-only players as compared to modern multi-format players because they are better made, start up/play much faster and the drawer mechanism is also much faster. You don’t have to wait for the device to check the format, read the contents, etc before it starts playing. If you click “eject”, it immediately ejects; no need to wait for the device to do who knows what before it ejects the disc.
To reduce overall system power consumption, many of today’s high-speed logic devices (e.g. FPGAs) operate from supplies of 3V or less. Lower bias voltages (and consequently lower logic thresholds) complicate interface with 5 V devices, creating a need for a fast and robust logic level shifter… the Si86xx isolator (can be) used as a logic threshold level shifter where each side of the isolator is biased to match the local logic rails. Note the common ground on both sides of the isolator since alllogic supplies are assumed to be connected to a common ground.
However, I don’t have to assume that the logic supplies are connected to a common ground and instead I am going to isolate the two grounds as much as possible by powering the input side with a completely separate supply, including using a separate AC transformer.
I salvaged another transformer (this one gives 9V DC unloaded) and a 7805 series LDO regulator
I had to rearrange the supplies in order to fit them in the chassis. This arrangement is even better as the line voltage components and wires are confined to the left-most area of the case and as far as possible from the low voltage electronics.
FULL GROUND ISOLATION
To eliminate the ground loop I was experiencing with the external RCA connectors I had to isolate the RCA connectors from the chassis. That did not explain the source of the ground loop though, it merely eliminate the additional path to signal ground. I had no supply ground wires connected to chassis and the only ground connection to chassis was the safety EARTH ground, so where did the ground loop come from?
After some poking with a voltmeter, I figured the source of the ground loop. The mounting holes of the DAC board are encircled with ground pads. These short to the mounting posts and the mounting posts short to the chassis.
If you don’t have a second path (for the signal) to ground, then you don’t have a ground loop. Having signal ground connected to the chassis is probably a safety feature but I have built all my electronics with the signal ground (the supply ground) isolated from the chassis and for safety Iuse a 3-wire power cord with the EARTH ground connected to the chassis. If using the chassis to shield sensitive electronics, then it is a good idea to not share the chassis ground with the electronics ground.
To isolate the mounting posts, I used poly washers for the mounting screws and put them on the top side and bottom side of the board:
These were made from presentation transparencies (remember those?) with a regular 3-hole paper punch and a smaller craft punch. The came out very nice.
Please also refer to:
Here is my board (S/N 000003) and even personally signed by Soren :-). I will document my build in this post.
Just finished building a +/- 12V power supply and had it powered-on an entire day in order to ensure that nothing was wrong with it. Most of the components are from my “pile of electronics”: some now, some recycled.
The transformers were taken from “surplus” unregulated wall supplies and are rated at 15V AC (which are too high of a voltage to use directly on the DAC board).
It is as basic and standard as a linear supply could be.
The AC rectifying part is basically the dam1021 input section
- Two transformers (AC 15V)
- Single bridge (2A)
- Smoothing capacitors. In this case 35V, 2200 uF standard Nichicons, one per rail
The film caps are additional bypass close to the regulators.
The “hardest” part of the build was manually placing the components on the board for a good fit. Maybe a CAD layout tool would had been a great help.
Not very pretty, but works fine. The thicker wire in the center is GND.
If building the PS from scratch is too much of a hassle, there are many kits from eBay for very little money. (You still need to source the transformers). The most popular ones are the adjustable ones based on the LM317/LM337. One good example is the following kit [link].
POWER SUPPLY MODS
For now, I am not thinking about PS mods, but here is a post with a lot of good information and measurement on the DAC in general and particularly on the power section [link].
After reading the post, these are some of the things that come to mind:
Reducing Power On/Off POP
During Power off, the positive rail drops faster than the negative rail (remember that the positive power consumption is 3X negative power consumption) and this creates a huge POP when the device is practically powered by the negative rail. – Using the additional capacitance of the regulated supply would probably make this problem worse because the capacitors are equal in size between the positive rail and negative rail.
The left LED is on the positive supply. It goes off first when cutting the external power
- Increase the power draw of the negative supply to be equal of the positive supply: you can do this by adding resistors to ground on the output of the supply.
- Adjust the amount of capacitance between the two rails so that during power off, the voltages on the two rails decay in approximately equal rate. This may or may not work, but it sounds that it could.
- Separate the analog supply (opamp and voltage reference) from the digital supply which only uses the positive rail. This requires major surgery of the board.
Snubbers on the transformer
The shouty sound can be somewhat tamed by filtering the power line and using optimal snubbers for the power transformer. (Check the Quasimodo/Cheapomodo threads for an excellent snubber measurement jig by Mark Johnson.)
Bypass bridge rectifier (this also “skips” the smoothing capacitors)
Update: I took a closer look at the J2 connections on the backside and the board and the +/- analog power connections are connected to the power lines after the RC filter. If you power through J2, there will be a 12 ohm resistor to the smoothing caps. This would not destroy the board, but not the right design. (It is actually safe [link]). This is probably only useful if you have a well filtered and regulated DC supply and you are operating near the headroom required by the 5V regulator which is around 7V. In any case, it is better to power through J1.
If using a regulated supply for input power, it is possible bypass the built-in bridge rectifier which is used for AC input. For DC input it is basically serves no function except it adds extra protection in case you inadvertently apply the wrong polarity. I measured the PWR A+ and PWR A- connections in J2 against the + and – poles of the input filter caps and I measure continuity. It would be better to use the GND connections of the power input (the ones in J1)) since they provide a solid and hefty connection to the GND plane. The ground connections on J2 are through thin traces.
But I need to respect the manufacturer’s warning, so here it goes [link]:
I can only recommend to supply any power on J1, the diode bridge used on the input is a low noise schottky type.
J2 is NOT for supplying power, it’s for testing or for sourcing small amount of power for external input circuits. Applying power will probably blow the board.
But on the other hand, just go ahead, I’ll be happy to sell you a couple of new boards :-)
Connecting the input power this way, bypasses the bridge rectifier (and skips the smoothing capacitors. I say “skip” because it does not bypass them since they are still connected through a 220 ohm resistor).
Replacing and adding on-board PS capacitor
This one is endorsed by Soren [link]
If you insist to improve the on-board power supply, try replacing the 6 electrolytic capacitors with aluminum polymer types, 1000u 16V exist in same 10mm SMD footprint and t.ex. digikey stock them at $2.20 each. Should be easy to replace.
You can also add a small polymer electrolytic on the 3.3V output, but please note that the clock oscillator power already have a filter, so I doubt it will make any difference.
Here is the 3.3V regulator. You may add a capacitor between Vout and GND (pins 1 adn 2)
At this time I would be doing a standard installation before thinking of any other mod.
Old electronics are excellent for these projects. Not only they are free, but at the minimum you get athe power cord/socket installed and some even come with power filters.
This one has a power socket for a detachable power cord and ground safety (the green wire)
Lots of useless buttons, but not bad looking at all :-)
Reusing the RCA connectors: one set for single-ended raw output and one set of single-ended buffered output
I like to use the analog-audio cable assemblies found inside (older) PCs. They have three leads (for left, right and ground) and are shielded.
It is recommended to use a 10K pot. (although a different value one might work as I beleive it is measuring voltage)
I am using one from a gutted Sony analog surround processor (when Japan used to make stuff :-)) which happens to be 10K potentiometer.
I will be using what I think is the best Toslink Module: the original one from TwistedPear Audio. This one has the Toshiba TORX142 module (25Mb) with supports up 192KHz sample rate (data sheet: torx142l), but Toshiba stopped making them. I purchased this several years back with the OPUS DAC [link] (which got me started in this audio DIY thing). The current one that is on sale at the TwistedPear Audio store is specified to support up to 96KHz [link].
The module has an integrated 3.3V regulator [link] and thus it is compatible with the 3.3V input limit of the R-2R DAC. In addition, the regulator is specified to accept 5-12V DC (with a spec max of 18V). It can therefore be powered directly by the 12V supply. No need to pull the power from the DAC.
The case even has a cut-out for the Toslink module…
USB to I2S Module
I am going to use the original DIYINHK XMOS-based module [link]. This one is USB powered.
Wanted to first check out the DAC with a basic and default configuration: Toslink Input, volume control, and SE buffered output to headphones.
Power connections, volume control connections and SPDIF connection.
Single-ended buffered output to a front-panel headphone jack.
The output opamp, the LME49724 [link] can drive a high impedance headphone directly. It is specified to drive a 600 ohm load meeting full specification.
Since the Senn HD-580 I use has an impedance of 300 ohm, I expect only a slight deviation from the specification, if any.
Front panel: power LED (in the power button), toggle power switch, headphone jack and volume control
I also installed an RCA jack for coax SDPIF. The Toslink module is powered by the +12V supply. The USB-I2S is powered by USB.
Back panel: SPDIF Coax, SE Raw output, SPDIF Toslink and USB-I2S
Just powered it up, original filters. Denon Multiformat player, Toslink output. Senn HD-580 (300 ohm) and Fidelio X1 (30 ohm).
First there was no sound either from the raw outputs or the SE buffered outputs.
- The signal lock LED was steady. In auto-input mode, the signal locks quite fast, as it should
- The voltages on J2 were all correct
- Removed the volume potentiometer connections, leaving just the SPDIF input
- Double checked all the output connections
It turned out that I had soldered the wrong connectors on the RCA jacks and the Headphone jacks!
Signal lock LED
- Steady on: signal lock
- Blinking: no signal or no lock
Then there was a hum coming from the raw output connected to the RCA jacks (RCA-Head Amp-Headphones). I figure there got to be a ground loop. I measured continuity between the chassis and the RCA GND.
Had to remove the metal plate that was grounding the outer sleeve of the RCA jacks to the chassis (the little metal tabs on each RCA jack).
Volume control rustling/crackling noise
I hear a faint static-like click when adjusting the volume. It is like the old analog amps when the pot was dirty or worn out. But this is digital. I shall investigate this further…
I added a 0.1 uF capacitor between 3.3V and Gnd and between the wiper and Gnd. No improvement. I think the problem could be in software (the FPGA volume software)
HOW DOES IT SOUND
Using Toslink source from Denon multiformat player, single-ended buffered output to a Senn HD-580 headphone. Compared with the analog output of the same player through a Fiio top of the line headphone amp.
The immediate most noticeable thing in an A/B comparison with the Denon player is the larger soundstage of the R-2R DAC. Further listening indeed shows a more expansive soundstage, like each instrument and voice fills more of the space around it.
The bass is also more “full bodied” and more impactful. Beautiful bass. I am using the original filter and do not detect any “harshness” as others have.
Perhaps comparing with the Denon is not such a good comparison, but it was easy and shows the that DAC performs very well.
I shall try other filters. Perhaps then I would feel the original filter results in some harshness…
Also, with direct output, I prefer the HD-580 to the Fidelio X1.
GOT THE I2S GOING
Finally connected the DIYINHK XMOS I2S module (this is the original model, different from what Soren is using -I believe the current isolated one).
Listening with iTunes and having iTunes upsample everything to 176.4KHz. Sounds good…
The DAC automatically selects the active input right away. On the XMOS device, once you apply power, there is bitclock and the DAC locks to it.
I am still using original firmware, so the slight clicks when changing sample rate are there. The rustling/crackling sound of adjusting the volume is also still there. I am sure these will be fixed, it the meantime, they don’t bother the listening experience.
In order to power the input side of the isolator, I had to build a small supply with a 5V regulator and a 3.3V regulator (didn’t have anything else suitable). The 3.3V regulator is a surface mount device so I had to solder legs and small heatsink tab. Just a crude but effective job :-)
How does it sound?
Whereas in the previous comparison, the DAC showed a more expansive soundstage, and the bass was more “full bodied”, more impactful, using I2S input (Windows 7 laptop, iTunes 12 with upsampling to
176.4KHz 192KHz [Check my post on adjusting the iTunes playback sample rate [link] and diyinhk XMOS interface) results in another step forward towards better sound and a enjoyable experience. This time the improvement is presented as music with more clarity or “crispiness” (more details?) as though the instruments give out a more “defined” sound. I think this reviewer gives a pretty good description of what I want to say: [link].
The Soekris R2R DAC is finally out, officially named
The Soekris dam1021 Sign Magnitude R-2R DAC
I will be collecting useful information here because as the thread at diyaudio gets longer, it would be difficult to find things…
The initial release (“v1) of the firmware together enables the following features:
- I2S input up to 384KHz sample rate (tested to 192KHz)
- SPDIF input up to 192KHz (tested to 96KHz)
- Automatic De-emphasis for 44.1KHz material
- Built-in set of simple FIR filters for all sample rates
- Digital volume control through simple potentiometer
- Automatic input selection
- Data reclocking: s/w PLL with 0.02 Hz low pass filter
- S/W interface (serial interface) allows:
- Volume control (e.g. with Arduino)
- Input selection (e.g. with Arduino)
- Loadable FIR filters including bypass filter for NOS support (s/w utility included)
- Firmware update/upgrade
Upcoming features (through user-performed firmware upgrade)
- DSD support
- Filter cascading for digital crossover
- Filter cascading for balanced use
- Master clock output
- 4 FIR1 filters for each sample rate selectable through the serial port with a default selection (#2)
- Linear Phase
- Medium, optimized mix between Linear Phase and Minimum Phase
- Soft, mostly Minimum Phase but not that good alias rejection
- No filter, also called non oversampling, no alias rejection
- Improved response for FIR2 filter which will be modified to be pretty soft….
- Enabled serial port on the isolated side (3.3v)
DAC CONNECTIONS 
SPDIF Input Connections 
The transformer for coax input is 1:1
Note: You can take the 1.2v or 3.3v from another supply.
I2S Input Connections
- Connect BCLK to I2S BCLK IN Pin
- Connect LRCK to I2S LRCK IN Pin
- Connect Data to I2S DAT IN Pin
- Provide external 3.3V to ISO +3.3V Pin
- AND connect I2S GND to ISO GND
The isolators are powered from both sides: a “dirty” side and a “clean” side. The power for the “dirty” or input side is provided by the input side electronics (your USB device for example)
Volume control connection [link]
Requires the use of a 10K linear potentiometer. Volume can be controlled -90 to +15 db
- Connect low side of potentiometer to GND Pin (left pin when viewed from front)
- Connect high side of potentiometer to +3.3V Pin (right pin when viewed from front)
- Connect Volume Pot wiper to VOLUME POT Pin (center pin when viewed from front)
If nothing is connected, the volume level defaults to 0 db.
The advantage of using a potentiometer, as compared to using a rotary encoder (which at the moment is not supported anyway), the DAC always starts with the last volume setting.
|Input|| INPSLCT0 Pin
|| INPSLCT1 Pin
|Auto Selection||Open||Open||DAC will search the 3 inputs for a valid signal and lock when found|
|I2S||GND||GND||Be aware that even when not used the USB-I2S interface might output a clock that the dam1021 lock on to….|
|SPDIF 1||Open||GND||Sensitive LVDS Receiver -used for Coax |
|SPDIF 2||GND||Open||Standard 3.3V digital level -used for Toslink|
For uP control, firmware update, etc. there is an RS232 serial port which is just like a PC serial port
- Pins are: RS232_RXD, RS232_TXD and RS232_GND
- Connect at 115200, 8, n, 1.
If your computer does not have a serial output, you might need a USB to serial cable such as this [link]
Notes on serial port 
- Best is running directly to a real PC serial port, but those are getting rare nowadays.
- The serial port transceiver on the dam1021 R-2R DAC has power savings enabled, to reduce noise. Tt need valid RS-232 level on the RXD line to power up. If your USB-Serial dongle also have power saving then you have a problem….
- 115200 is kinda pretty fast for RS-232, make sure you have a good but not too long cable, and if using a USB-Serial dongle, check if it can actually run 115200.
The RS232 serial port is enabled by the Intersil 3221ECVZ chip [link]
Master Clock Output Pin 
- I2S MCLK OUT pin: Master clock output: 45.1584 and 49.152 Mhz (which can also be divided)
- I2S FSEL IN pin: Selects between 45.1584 MHz and 49.152 MHz
Same master clock output is also used when connecting multiple DAC’s, for example for digital crossover applications.
Power Rails Connector/Header 
J2 is a connector with all the power rails, for testing or for supplying (limited) power to other things. There is 1.2V that can be used to bias the LVDS for Coax SPDIF input
Note: in the diagram, the signal polarity is correct, but the left/right channel is reversed 
I think I have the channel assignment correct in the photo below. (If you install the board with the XLR connectors in a back panel, the left channel would be on the left side and the right channel on the right side)
Single Ended Output Connection
You may choose the raw output or the buffered output. The raw output comes straight from the resistor ladder through a low pass RC filter (625R Zout, capacitor is 1200 pf ceramic NP0, -3db is then at 212 Khz. NP0/C0G types are supposed to be fine, especially when their effect are way out of the audio band…. [link]). The buffered output is out of the opamps (schematic below).
The only choice for balanced output is buffered output from the opamps. There is already a footprint to install the included XLR connectors or you can use wires to connect to a panel mount XLR jack or directly to a XLR cable.
Output Buffer Schematic
(This is an updated information from my previous post: R-2R DAC FOR THE REST OF US [link])
Main Power Section
- Designed to be powered by a single dual 7-8V AC, 5W transformer (since it has has a bridge rectifier installed)
- Can also take an external +/- 7-15V DC supply. (See section above on power supply requirements for further details)
- Actual power consumption has been measured to be 2.4W.
- Filter capacitors are Nichicon 820uF 16V CL series [link]
- Negative voltage is required for the output opamps and other parts of the circuit [link]
- A DC-DC converter (switch mode) provides the 1.2V for the FPGA core. Every other supply is low noise linear [link]
- “The LME output buffers are powered via an additional large RC filter after the main capacitors, no active regulators. With a typical PSRR of 125 db I didn’t worry much about 100/120 hz ripple, only worried about higher frequency noise on the power rails….”
Input Voltage 
- DC: +/- 7 to +/-15V DC; preferable 9-12V DC
- AC: 2x 7-8V AC
- Power goes though a diode bridge so polarity doesn’t matter. Connector is MTA156 type.
Maximum and Minimum Input Voltage 
- Upper limit of 16.5V, based on capacitor voltage and also to limit loss in linear regulators.
- Lower limit of 7.5V, based on loss though diode bridge and 5V linear regulators.
- Taking into account line voltage tolerance and transformer no/low load voltage, this results in the 7-8V AC requirement for transformers.
Power Consumption 
- Positive Rail: .18A, 10V
- Negative Rail: 0.06A, 10V
- Total: 2.4W
- The positive supply draw about 3 times as much current as the negative; the current is almost independent of input voltage.
A 3.3V LDO Powers the Clock [link]
It is a pretty hefty regulator. And it seems the only 3.3v regulator on the board. It must also supply 3.3V to:
- 3.3V needs to the FPGA
- Clean side of signal isolators
- SPDIF LVDS receivers
- Flash memory
- Other components (like the shift registers?)
Good thing it is implemented next to the clock of all places.
Reference Voltage Supply
The most critical supply is the +/- 4V reference for the resistor ladder. This is generated by a “two step, first to +- 5V (by linear regulators), then to +-4V by precision low noise medium current opamps”; “-4V reference is sent though an inverter with 0.01% resistors generating the +4 reference”. The references are further “filtered and buffered for each rail and channel”.
The 8L05A and 9L05A are +5 and -5 Linear regulators [link]. Input to these regulators are the +/- rails (which could be 7 to 15V regulated or unregulated) . The outputs are filterd by 100uF capacitors.
DIGITAL FILTERS 
Original (current) digital filters:
- FIR1, upsampling from incoming sample rate to 352/384 KHz in one step, with different filter length based on incoming sample rate. All FIR1 filters are basic Parks-McClellan “brick-wall” types, designed with http://t-filter.appspot.com/fir/index.html, but still shorter than your regular DAC.
- FIR1 is automatically bypassed if feeding 352/384 KHz data.
- IIR, bank of 15 bi-quads operating at 352/384 KHz, with one used for the CD de-emphasis filter, none otherwise used for the basic DAC.
- FIR2, upsampling from 352/384 KHzto 2.8/3.1 MHz, reasonable short and soft but still using same design as FIR1.
- All filters are using 32 bit coefficients, with up to 67 bit MAC accumulator.
I’m not a believer in no filters (non oversampling), but also don’t like the sharp “brick-wall” filter types with the pre-ringing. The goal is to work towards filter types that remove just enough to not cause problems with aliasing. It’s pretty easy with higher sampling rates, but is long and hard work and listening tests with 44.1 KHz, which still are the sample rate mostly used….
Further details on the filters [link]
FIR1 is operating at 352.8K/384K (output frequency of the filter and this is set in hardware [link]) and each filter can have up to:
- 1016 tabs at 44.1K/48K input sample rate
- 508 tabs at 88.2K/96K input sample rate
- 252 tabs at 176.4K/192K input sample rate
- 124 tabs at 352.8K/384K input sample rate, but normally bypassed
This also means that:
- 44.1K/48K input sample rate, the oversampling filter must be 8X
- 88.2K/96K input sample rate, the oversampling filter must be 4X
- 176.4K/192K input sample rate, the oversampling filter must be 2X
- 352.8K/384K input sample rate, , the oversampling filter must be 1X, but normally bypassed
All upsampling is done by zero insertion, therefore gain needs to be set to match oversampling rate.
- For 44.1KHz/48KHz sample rate the upsampling is 8X, thus the gain should be 8
- For 88.2KHz/96KHz sample rate the upsampling is 4X, thus the gain should be 4
- For 176.4KHz/192KHz sample rate the upsampling is 2X, thus the gain should be 2
- For 352.8KHz/384KHz sample rate the upsampling is 1X, thus the gain should be 1 (But this filter is normally bypassed
Headroom for clipping prevention due to oversampling and the required applied gain [link]
The dam1021 have 2-4 bit of headroom though the digital filters, all the way until the volume control. So any clipping would be because of incorrect filters.
Why not one step oversampling to 2.8/3.1 MHz? 
Not that practical, would require > 3000 taps. In addition, the intermediate frequency is good to have for the CD de-emphasis and crossover filters.
352K/384K is as a good compromise for the intermediate rate, so there is space for enough IIR filters for crossover use and room correction. If you go up you get less IIR filters. # filters = 45M/49M divided with intermediate rate divided with 8 = 16.
FIR2 is operating at 2.822M/3.072M and can have up to 120 tabs, with input sample rates 352.8K/383K.
The FIR2 filter don’t require much, I’m looking into doing a bessel or butterworth type filter there.
dam1021 FPGA fixed point format [link]
The filter file 1021filt.txt
There can be multiple of each filters in the 1021filt.txt filter file (part of the filter tools), but at least one (filter) needs to be there for each input sample rate. Currently just the first one for a given sample rate is used, in later firmware releases you will be able to choose between different filters
Filter tools and documentation will be shortly available for users to upload custom filters to the FPGA. The tools are already available here: [link]
UPDATING THE FIRMWARE 
- Download and unzip attached file 1021fpga_090.zip
- Connect the dam1021 serial port to a PC serial port with a terminal program (I use HyperTerminal), set for 115200,n,8,1, no handshake. (DAC must be powered up)
- Enter uManager on dam1021 by typing “+++” followed by a 1 second pause.
- Type “download” and start sending file from the terminal program using 1K X-modem protocol.
- Power cycle when done, you can verify by entering uManager again, FPGA revision should then be 0.9.
Great that the firmware can be user-upgradable. Using standard serial port makes it even more convenient.
Further details here:
R-2R DAC For The REST of US [link]
Picked up a huge Marantz receiver (SR8200) at the local donation center for not very much money.
FAULTY VOLUME KNOB
Even though the donation center has a 7-day return policy for non-working electronics, the receiver was worth more than I paid (to me) in parts alone. I quickly discovered that the volume knob was “stuck”. It is a rotary encoder type (not a potentiometer type). The volume setting would barely move when turning the knob,
Thanks to my familiarity with rotary encoders, I quickly recognized this problem as “noisy transitions” within the rotary encoder. In other words, it needed (more) debouncing. What I did was to install some capacitors to the signal pins and viola! it works almost as new. There is still a bit of debouncing problem but does not affect the responsiveness of the rotary encoder. If I experiment with different value capacitors, I would likely solve the problem, but for now this is good enough.
Other than this, the unit seems to be working properly. The only disadvantage is that now I cannot justify gutting it for parts :-)
From the golden era of Made in Japan audio electronics. Things are put together with more screws than seemingly necessary. Plus, this is the first device where I find the use of copper (or some copper allow) screws. The chassis is made of traditional stamped steel.
Nice brushed aluminum front panel (but the knobs are “metal looking” plastic)
The most ELNA capacitors in one place!
One of the last through-hole, hand-crafted audio components…
This receiver, old enough to be powered by a liner supply, is rated at 6x130W (780W for the amplifier section).
It uses a large EI transformer with a copper flux band. These bands are used In order to reduce the radiated flux from the transformer core, acting as a shorted turn to the leakage flux (only), greatly reducing magnetic interference to adjacent equipment.
There are two 27,000 uF “Marantz” filter capacitors. Incredibly good looking! I believe they are made by ELNA (as every other capacitor is also ELNA). The heatsink behind the capactors is for the bridge rectifier. .
(Update: a reader alerted me that the caps are made by Nippon Chemicon. The logo is in plain sight)
There is space for two additional capacitors. A nice mod would be to add a couple of Panasonic 4-lead capacitors such as these: Panasonic T-HA 10,000-18,000 uF, 63V [link][link] (with care not to blow the fuse due to in-rush current during power-up)
The SR9200 uses 4 capacitors with higher voltage rating but lower capacity as shown in the photo below [link]
ANALOG VOLUME CONTROL
The volume control is provided by two 6-channel Toshiba TC9482N volume control [link]
These devices control up to 8 analog channels (7.1 multichannel) that area available as pre-out but only 6 of them connect to power amplifiers
The input and outputs are buffered by NJM 2068DD opamps [link]. The “DD” grade devices exhibit lower noise specification. Where have we heard about these NJM2068?… From the development of the famous O2 headphone amp [link]
BOTTOM LINE: For those wanting to skip the Tech Section, the conclusions can be summed up as follows:
At gains less than 4X nothing overall could beat the $0.39 NJM2068 in the O2’s gain stage. This is especially true if you’re concerned about power consumption for battery operation.
Current prices of the 2068DD are $0.60 in quantity 1 orders [link]
DIGITAL TO ANALOG BOARD
Stereo D/A board uses CS4396 D/A (3 of them).
A 6-channel module with forced cooling.
Local power supply bypass capacitors. Notice the space for larger size capacitors (the higher model SR9200 uses larger capacitors). Replacing these capacitors with larger ones (a 1000 uF nichicon KW [link] for example -maximum diameter is 16mm) would be an easy mod.
Local PS bypass capacitors in the SR9200
DIRECT AUDIO PATH
There is an 8-channel analog input option (7.1 input) that bypasses all the digital processing. They are controlled by the analog volume chips and the output is available through the 8-channel pre-out. Six of those 8 channels are connected to the 6-channel power amplification module. This receiver can be used as a stereo tri-amp setup.
(12/22/14- Updated with information from AKM support engineers -see register section)
It has been a long time semiconductor houses invested in a flagship product. Wolfson announced the WM8471 in 2007 and ESS announced the Sabre DAC in 2008. Recent investment has been concentrated in DACs for the broad consumer industry especially for the mobile segment. It is good to see a company still interested in investing resources for the “audiophile” segment.
AMK introduced the AK4490 this year and has recently made it available in production quantities. It differs upon the AK 4399 DAC in the following areas (yes, the spec for Dynamic Range is lower in the new chip):
|THD||-112 dB||-105 dB|
|S/N (Mono)||123 dB||126 dB|
|Max Sample Rate||768KHz||216 KHz|
|Built-in Digital Filters||5||2|
|Direct DSD (No conversion to PCM)||Yes||No|
|AVDD Max operating voltage||7.2V||5.25V|
Here is an overlay of the FTT measurement between the AK4490 and AK4399 (graph slightly shifted to the right to show the comparison) from the evaluation board data sheets. As seen, the AK4490 has a slight edge over the AK4399:
Increasing S/N by 3 dB
In order to “recover” the lost S/N in the new device as compared with the old device, The AK4490 can be operated with an analog supply of up to 7.2V. At 7V we gain 3dB S/N resulting in 126 dB for mono operation and therefore meeting the best specification of the old A4399 part.
Even though this is not documented in the current version of the AK4490 data sheet, it is documented in the AK4495 data sheet:
Thus one of the “mods” that can be made in this DAC is to run the DAC at the higher-end of the analog voltage operating spectrum.
Built-in Digital Filters
(images taken from Ayre’s paper [link]):
The built-in digital filters consist of 5 selectable filters. They include all the “popular” filters developed so far by different vendors plus one additional filter with undisclosed response (super slow roll-off). The filters are described as follows:
Linear phase Sharp Roll-off (AKM notation: “no delay”): this is the “standard” sharp roll-off filter found is all DACs. It is also known as the “brickwall” filter. It is said that pre-ringing sounds unnatural.
Linear phase Slow Roll-off (AKM notation: “no delay”): this is also a “standard” filter found in all DACs. As in the linear phase sharp roll-off filter, it also generates pre-ringing, but trading lower amounts of pre-ringing with letting more aliased image through (theoretically increasing harmonic distortion).
Minimum delay Sharp Roll-off (AKM notation: “short delay”): this is also called the “minimum phase” or “apodizing” filter that was the rage a few years back. Whereas in the past audio engineers have insisted in phase linearity (meaning all frequencies have equal phase or delay), More recent research have shown that a “minimum phase” filter sacrifices some of the phase linearity (adds some phase distortion) for better time response. This filter removes all the “unnatural” pre-ringing but “dumps” all that energy to post-ringing. Implementation of this filter is also found in the Wolfson WM8741/8742 DACs
Minimum delay Slow Roll-off (AKM notation: “short delay”): this is a “more modern” type of filter also found in the Wolfson WM8741/8742 DACs. In addition to eliminating pre-ringing, this filter also incorporates slow roll-off and this reduces post ringing as well.
The properties of this filter are similar to the “MP filter” found in Ayres latest CD player.
Super Slow Roll-off: this filter is the differentiating feature (in terms of built-in filters) that this DAC provides. The AKM literature says “super slow roll-off filter with emphasized characteristics” (which really means nothing). There is some information in the marketing page as shown below.
The marketing information says the following [link]
Native DSD Support
Supports 2.8MHz (64fs), 5.6MHz (126fs) and 11.2MHz (256fs) DSD
According to AKM, the volume control module and the delta-sigma modulator can be bypassed for DSD resulting in “direct” DSD rendering. The AK4490 contains an integrated low-pass filter specifically for DSD data. The ultimate specified performance for SACD (as described in the Scarlet Book) can be easily realized with a simple external analog filter.
Notice the bypass path for DSD Data. The DSD data is received by the DSD interface and sent directly to the “SCF” (Switched Capacitor Filter) block. DSD filter can be selected at 50KHz, 100KHz or 150KHz cut-off.
Other Comparative Features
Resolution32 bit32 bit32 bit24 bit24 bit
|DR (Mono)||123 dB||135 dB||127 dB||128 dB||132 dB|
|THD||-112 dB||-120 dB||-120 dB||-100 dB||-108 dB|
|Output Mode||Voltage||V or I (best)||V or I (best)||Voltage||Current|
|Resolution||32 bit||32 bit||32 bit||24 bit||24 bit|
|DSD Mode||DSD Direct and DSD to PCM||DSD to PCM||DSD to PCM||DSD Direct and DSD to PCM|
Just like the WM8741, the AK4490 supports “direct DSD” processing bypassing the volume control and delta-sigma modulator. And like the WM8741, there is no automatic switching between PCM and DSD.
I2S and DSD shared lines
In order to facilitate the playing of both PCM and DSD content, it is desirable to have the same lines transmit PCM and DSD data. We find that in the AK4490, the I2S and DSD signals are shared. Here is a post I write earlier concerning shared I2S/DSD signal lines: [link]
The table below shows compatible DACs (DACs that share that use the same lines for DSD and PCM) and interfaces showing how the DSD pins are mapped to the PCM/I2S pins:
||XMOS Ref [link]|
|BCLK||DSD Clock||DSD Clock||DSD Clock||DSD Clock||DSD Clock||DSD Clock|
|LRCLK||DATA Left||DATA Right||DATA Right||DATA Left||DATA Left||DATA Left|
|DATA||DATA Right||Data Left||Data Left||Data Right||DATA Right||DATA Right|
The AK4490 DAC follows the mapping of the AK4399 which switches channels with the “conventional” channel mapping of USB interfaces. Likely it was the USB interface designers that took notice of the ESS9018 DAC and conformed the channel mapping to that chip.
Fortunately, there is channel remapping in at least the Amanero interface and there is channel remapping in the DAC itself as specified in the following table of the data sheet:
MONO=0, SELLR=1 says:
- Right channel input is mapped to Left channel output
- Left channel input is mapped to Right channel output
I Just received diyinhk’s implementation of AKM’s new flagship DAC, the AKM AK4490EQ [link]. This is the first available diy board in the market (that I know of):
POWER SUPPLY LINES
The Diyinhk implementation follows (mostly) the AKM evaluation board and data sheet [link] but maximizes performance whenever possible (like in the selection of capacitor type and value). The board is powered by: 5V line, 3.3V line and +/- 12V line (for the output opamp).
The general layout of the power traces, decoupling capacitors and ground planes also follows the data sheet:
Grounding and Power Supply Decoupling:
To minimize coupling by digital noise, decoupling capacitors should be connected to AVDD and DVDD respectively. VREFHL/R and VDDL/R are supplied from analog supply in system, and AVDD and DVDD are supplied from digital supply in system. Power lines of VREFHL/R and VDDL/R should be distributed separately from the point with low impedance of regulator etc. AVSS, DVSS, VSSL and VSSR must be connected to the same analog ground plane. Decoupling capacitors for high frequency should be placed as near as possible to the supply pin.
Analog 5V supply lines (can operate up to 7.2V according to spec)
The 5V supply connects to VDD (5V Analog supply input) and Reference Voltage High (VREFH) -as recommended in the data sheet.
The differential voltage between VREFH-L/R and VREFL-L/R sets the analog output range. The VREFH-L/R pin is normally connected to VDD (analog 5V supply), and the VREFL-L/R pin is normally connected to VSS1/2/3 (analog ground). VREFH-L/R and VREFL-L/R should be connected with a 0.1µF ceramic capacitor as near as possible to the pin to eliminate the effects of high frequency noise…All signals, especially clocks, should be kept away from the VREFH-L/R and VREFL-L/R pins in order to avoid unwanted noise coupling into the AK4490.
In addition, according to the eval board manual, a large value capacitor between VREFH-L/R (Analog 5v) and VREFL-L/R (GND) improves the THD performance in accordance to the following graph:
The Diyinhk board is implemented with 2200 uF capacitors, achieving the best THD numbers. (The larger capacitor holds the reverence voltage stable -perhaps an even larger capacitor would further improve the low frequency THD numbers).
There is an option to use separate supplies for right and left VREF and VDD. This also follows the scheme implemented in the official evaluation board where the left VREF is separately powered from the right VREF.
Further, the AKM literature states:
Special designs techniques for sound quality are applied to each blocks for achieving balanced, smooth and powerful signal flow. In addition to L/R perfectly symmetrical layout, more than 5x trace width is used for signal line compared existing products, supplying rich current to analog signal output blocks. To achieve low impedance, two analog power supply pins and two signal reference pins are assigned for each channel, allowing the system to utilize thick PCB trace pattern giving low impedance sources.
The board takes advantage of this feature to use thicker lines for VREF and VDD
All 0.1 uF decoupling ceramic capacitors are C0G
The official evaluation board has a provision to separate the VREF from the Analog 5V VDD which is not implemented in this board. However, it is easy to mod and use separate supplies for VREF and Analog 5V VDD.
The evaluation board implements VREF with the following circuit:
3.3 V Supply Line (Analog 3.3V and Digital 3.3V)
There is a 3.3V analog supply pin and a 3.3V digital supply pin in the chip. The default implementation of the diyinhk board uses the same supply line but filters them with a ferrite bead. By removing the ferrite bead, the user can use separate supplies for the analog and digital 3.3V.
In the evaluation board, AVDD and DVDD are powered by separate regulators:
The ground planes follows the recommended separation between analog and digital sides (along pins 17-18 and 45-46)
The older device, the AK4399 supported a 3-wire serial interface. This seemed a not too widely supported protocol (it was not SPI and could not find a similar protocol in Arduino libraries , but one could code the protocol “by hand” as it was just a serial protocol -never tried it though)
Fortunately the new DAC supports I2C protocol (and maintains support for the original 3-wire serial interface found in older DACs). This greatly facilitating the interface to a microcontroller such as Arduino because of their built-in support for more standard protocols such as I2C and SPI.
The advantage of using the S/W interface is that it supports features such as volume control and DSD which are not available through the H/W interface.
The following table summarizes these features that are available in H/W interface (parallel interface -by pulling hardware pins up or down) and S/W interface (serial interface -microcontroller control).
Not indicated in the table is the “super slow roll-off” filter which is enabled by a register setting in s/w mode.
REGISTER DEFINITION SUMMARY
(Updated with information from AKM support engineer)
Here I summarize the register settings and the different functions that can be programmed. I also attempt to do some “translating” of AKM’s vocabulary to more “traditional” vocabulary.
I was able to communicate with AKM to clarify the functionality of certain sections.
Register address: 00 (Control 1) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| Reset chip without initializing registers |_|_|_|_|x|x|x|_| Interface mode: 16bit, 24bit, 32bit, I2S, LJ... (1) |_|_|x|_|_|_|_|_| External digital filter clock: 768KHz/384KHz |_|x|_|_|_|_|_|_| Enable/disable external digital filter mode |x|_|_|_|_|_|_|_| Master Clock frequency Setting: auto/manual (2)(3) NOTES: (1)- The only requirement for bitclock is >= 2x bit depth. Bitclock could be 32fs, 48fs or 64fs. Not limited to always be 64fs as in ESS DACs (2)- Auto: detects master clock frequency and sampling frequency (44.1KHz, 96KHz, ...) automatically; sets oversampling rate (1x, 2x, 4x...) according to input MCKL (this is kind of obvious). Note: AKM calls sample rate "sampling speed" and assigns names to typical sample rates: 44-48KHz="normal", 88-96KHz="double", 175-192KHz="quad"... (3)- Manual: manually set the sampling rate (44.1KHz, 96KHz...) Use reg 01 and reg 05 for sampling rate setting. This means, in its simplest form, manually matching the sampling rate to the incoming data sample rate to use the highest oversampling rate allowed by the system and thus obtain best noise performance. This feature can also be used to select a different sampling rate (typically a lower oversampling rate); for example, if selecting "normal" for 44.1KHz allows 8x oversampling (512fs), selecting "double" results in 4x oversampling (256fs). This allows for experimentation with different oversampling rates and can be used to tailor the sound for those inclined to lower oversampling or even no oversampling. The use of lower oversampling results in higher noise for these kind of DACs. AKM indicates in the datasheet that using a lower oversampling rate (512fs to 256fs) results in a decrease of S/N of 3dB. Register address: 01 (Control 2) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| Mute/unmute |_|_|_|_|_|x|x|_| De-emphasis: Off, 32KHz, 44.1KHz, 48KHz |_|_|_|x|x|_|_|_| Manual setting of sampling speed: "normal", "double"... (1) |_|_|x|_|_|_|_|_| Short Delay/Traditional filter (Minimum/Linear phase) |_|x|_|_|_|_|_|_| Zero data detect mode: Separate channels or ANDed channels |x|_|_|_|_|_|_|_| Zero data detect ON/OFF NOTES: (1)- Manual sampling speed setting uses 3 bits. The third bit is in reg 05. See notes on register 00 for additional info on manual settings Register address: 02 (Control 3) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| Filter cutoff slope: fast/slow |_|_|_|_|_|_|x|_| MONO mode: left/right |_|_|_|_|_|x|_|_| Invert output pin level on zero detect |_|_|_|_|x|_|_|_| MONO/STEREO mode |_|_|_|x|_|_|_|_| DSD Data on clock falling/rising edge |_|x|_|_|_|_|_|_| DSD master clock frequency:512KHz/768KHz |x|_|_|_|_|_|_|_| PCM/DSD mode Register address: 03 (Left Channel Attenuation) 7 6 5 4 3 2 1 0 |x|x|x|x|x|x|x|x| Attenuation (1) NOTES: (1)- 256 levels, 0.5 dB each. 00=mute; ff=max volume Register address: 04 (Right Channel Attenuation) 7 6 5 4 3 2 1 0 |x|x|x|x|x|x|x|x| Attenuation (1) NOTES: (1)- 256 levels, 0.5 dB each. 00= mute; ff= max volume Register address: 05 (Control 4) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| Super Slow filter on/off |_|_|_|_|_|_|x|_| Bit 3 of the manual sampling speed setting (see reg 01) |_|x|_|_|_|_|_|_| Left channel phase invert ON/OFF |x|_|_|_|_|_|_|_| Right channel phase invert ON/OFF Register address: 06 (control 5) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| DSD bit 0 of sampling speed selection (bit 1 is in reg 9)(1) |_|_|_|_|_|_|x|_| DSD Mode: Direct/Convert to PCM (2) |_|_|_|_|x|_|_|_| DSD Automute release when Automute release is in "hold" |_|_|_|x|_|_|_|_| Automute release: Auto/hold (3) |_|_|x|_|_|_|_|_| Right Channel DSD flag when detecting full scale signal |_|x|_|_|_|_|_|_| Left Channel DSD flag when detecting full scale signal |x|_|_|_|_|_|_|_| DSD AutoMute: ON/OFF (4) NOTES: (1)- There is no facility for setting auto sample rate detection for DSD. The use must detect the incoming DSD sample speed and match the sampling speed. Will have to experiment to see what is the effect of sample speed mismatch. (2)- In DSD direct mode, the volume control and delta-sigma modulator are bypassed. In PCM mode, it converts to PCM and uses volume control block and delta-sigma modulator. DSD direct with a combination of the internal filter and simple output filter meets the filter specification of the SACD Scarlet Book. (3)- Automute condition disappears when data becomes under full scale (4)- Automute condition is when data is full scale Register address: 07 (Control 6) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| Synchronize ON/OFF (1) NOTES: (1) Synchronizes multiple DACs when used together in the same system. Read data sheet for more information. Register address: 08 (Control 7) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|x|x| Sound Quality Control Setting (1) NOTES: (1): Sound Control has 3 settings: "1", "2", "3". The AK4495 data sheet shows additional settings "4" and "5". These setting refer to the 5 different filters that are available in the DAC. They serve the same function as the filter selection bits specified in the other registers. What is unclear is which register takes precedence. Register address: 09 (Control 8) 7 6 5 4 3 2 1 0 |_|_|_|_|_|_|_|x| DSD bit 1 of sample speed selection (see also reg 5) |_|_|_|_|_|_|x|_| DSD filter selection when in DSD direct mode
This should be the last info gathering post for building the First One Amp. Now I need to get a drill press in order to drill the holes on the heatsink…
The module is set to the correct operating parameters and tested at the factory. In case you need to check and readjust, most of the instructions can be found in this post [link] and following. The trimpots and test points are clearly maked on the board
OUTPUT BIAS CURRENT
The bias current when the amp is in idle (no input) is 220 mA for the output stage [link] (200 mA absolute minimum if you have heat problems [link]), plus 60 mA for the rest of the circuit. This is a fixed value regardless of the supply voltage [link]. Trimpot TR3 is used to set the output bias current.Thus:
- Output bias current = 280 mA. (minimum 260 mA)
How to measure output bias current
The simplest way to measure the output bias current is with a digital voltmeter in current measurement mode. Connect the meter in series with the positive power supply wire. Alternatively you can connect a low value (1 ohm) resistor in series with the positive supply wire and measure the voltage across the power resistor. [link]
DC OFFSET AND VAS BIAS CURRENT
The DC offset and VAS bias are set as follows:
- DC offset = 0v (+/- 10 mv)
- VAS bias = 12 mA when cold or 15 mA when idle for 20 minutes
How to adjust DC offset and VAS bias
TR1 and TR2 sets DC offset and VAS bias current at the same time. Both works in pairs reciprocally as best explained by this post from the VAAS thread [link]:
- Adjust (both together) TR1 and TR2 clockwise: increase VAS bias current
- Adjust (both together) TR1 and TR2 counter-clockwise: decrease VAS bias
How much to turn TR1 and TR2 depends on the DC offset, so you must adjust both to arrive at the correct VAS bias and zero DC offset. Using two DMM would make the adjustment easier.
How to measure DC offset
- To measure the offset short the amp’s inputs and measure DC at the outputs.[link]
How to measure VAS bias
- Amp cold: 12 mA bias: measure 120 mV between TP1(+) and TP2(-) or between TP3-TP4 (doesn’t matter which pair of test points).
- Amp idle for 20 minutes: 15 mA bias: measure 150 mV between TP1(+) and TP2(-) or between TP3-TP4
Here is a diagram of the 3 parameters that can be adjusted:
- Use TR3 to adjust DC Bias to 280 mA
- Use TR1 and TR2 to adjust DC Offset to 0 V and VAS current by measuring voltage of 120 mV (cold) or 150 mV (warm)
LC recommends the following ground wiring (basically a ground lift) for the amplifier modules [link]:
Improved schematic for stereo connection in a single chassis. GND potential of each channel is lifted from chassis-earth potential, meaning connection is done via 1 k resistor and anti-parallel diodes. In this way EARTH potential interference currents are isolated from GND potential. At the same time GND potential of each channel also isolated in between.
The purpose of the ground lift device (the diode-resistor-diode) is a compromise between best sound and safety [link]
Since I don’t want GND to be complete floating I tied GND from both channels to EARTH potential via a DRD (diode-resistor-diode) chain. It is a compromise needed for a safety reasons.
Best sound is at complete GND to chassis-earth isolation, so no DRD present. My demo amp, which I’m just listening at the moment, is in complete GND isolation (to EARTH).
The user is encouraged to install a switch that can short the ground lifting the device (the 1 k resistor and anti-parallel diodes) and experiment with both options [link]
You can even install GND lift switch on the back panel, shorting the DRD. So one position for GND (direct GND to EARTH connection) and another for GND lift connection (GND to DRD to EARTH connection)
So basically it is a “lifted ground” connection.
In the recommended hookup diagram above, the GND terminal of the Supply is connected to the GND terminal of the amp module. This is the obvious normal connection for proper operation. Each module has a single return path to the supply.
It is also necessary to connect the earth wire to the chassis for safety. This is to prevent exposing any harmful voltage in case a failure happens. Only if you have a “double insulated” chassis, then you can dispense with the earth connection (and this is what is called “Class II” appliance).
If one looks at the Hypex power supply specifications [link], they are built as safety Class II devices. This means they are already isolated with the minimum 6 mm from all possible conducting parts (its own metal frame). And can in theory be installed in a chassis without EARTH connection if you follow the double-isolation approach (meaning among other things that the wire you use for mains wiring has to be double insulated and having the expertise to double insulate everything else).
But in the normal approach of having an EARTH connected chassis, then the power supply’s metal frame becomes also connected to EARTH and with the 2-wire mains terminal, the power supply is in effect connected to the 3 mains wires in compliance with safety standards.
So from a safety point of view, signal GND connection to EARTH is not really a safety requirement (since the chassis is already connected to EARTH)
Now the question is “what is the purpose to connect the components GND terminal to EARTH?”
The answer might be in this application note from Hypex [link] where it says:
I can’t recommend separating the audio ground from the chassis ground, because that’s a recipe for making a radio receiver
So the reason is to prevent picking up electromagnetic radiation in the environment. And the best thing to prevent this is to connect the signal GND to the chassis EARTH.
The paper gives the following options:
- If you want to use RCA inputs, disconnect the mains earth and employ double insulated construction techniques.
- Use balanced (XLR) inputs. This allows the whole thing to be earthed unless the ancillary equipment has problems.
- Make a “pseudo-differential” RCA input. I still haven’t figured out whether or not I should post a detailed description of how to do this, because unless I manage to explain with perfect clarity it’s almost certain to generate large volumes of mail.
- Anything else (e.g. floating the amps inside a grounded chassis), but then you’re on your own if you hear your mobile through the speakers.
Thus connecting signal ground to EARTH it is about noise immunity.
Indeed, according to this article from RANE, signal GND must be connected to Chassis GND (which in our case, it is connected to EARTH ground) [link]
It is easy to confuse chassis ground and signal ground since they are usually connected together — either directly or through one of several passive schemes. The key to keeping an audio device immune from external noise sources is knowing where and how to connect signal ground to the chassis.
First let’s examine why they must be tied together… There are at least two reasons why one should connect signal ground and chassis ground together in a unit.
One reason is to decrease the effects of coupling electrostatic charge on the chassis and the internal circuitry. External noise sources can induce noise currents and electrostatic charge on a unit’s chassis. Noise currents induced into the cable shields also flow through the chassis — since the shields terminate (or should terminate) on the chassis. Since there is also coupling between the chassis and the internal circuitry, noise on the chassis can couple into the internal audio. This noise coupling can be minimized by connecting the signal ground to the chassis. This allows the entire grounding system to fluctuate with the noise, surprisingly providing a quiet system. Further coupling reduction is gained when the chassis is solidly bonded to a good earth ground — either through the line cord, through the rack rails or with an independent technical or protective ground conductor. This provides a non-audio return path for any externally induced noise.
The second reason to connect signal ground to chassis is the necessity to keep the signal grounds of two interconnected units at very nearly the same voltage potential. Doing so prevents the loss of system dynamic range where the incoming peak voltage levels exceed the power supply rails of the receiving unit.
WHY USE GROUND-LIFT?
According to this document on audio grounding [link]
Some people believe that it is necessary to isolate the system star ground from the chassis and safety ground in order to have a hum‐free audio system. However, if all of the components in the system have their grounding implemented properly, there is absolutely no need for ground isolation,
Although isolating the grounds may eliminate a ground loop, it does come with two penalties:
- First, since the signal reference (signal GND) is not directly connected to the chassis, the chassis is not an effective shield for the electronics
- Second, since the power common is isolated from the safety ground and connected to the signal reference, any AC leakage current from the power supply may flow through the signal reference to get to the safety ground in another component.
If you must isolate the grounds; never, ever, for any reason, disconnect a safety ground (chassis connection to EARTH ground) or fail to provide a safety ground in any equipment that you build. First, it is unsafe and second, there are equally effective methods of isolating grounds that do not come with the safety hazard. The following figure shows two such methods.
First is to provide a “ground lift” switch between the two grounds to be isolated…
A better solution is to provide a Safety Loop Breaker Circuit (SLB). This circuit will allow the current from a fault to flow to the chassis and also provide ground isolation under normal, non‐fault conditions.
You can find a circuit for a ground loop breaking (or SLB) from Elliot Sound Products [link] which is in principle similar to what LC is proposing. Thus the recommended circuit provided by LC is the proper method to avoid ground loops and be able to interface with upstream components with less than ideal ground implementations.
In my diy builds I’ve never connected the signal GND to EARTH, but have always connected EARTH to chassis. I think it would be a good idea to try connecting the signal GND to EARTH with a break switch to compare.
DOES GROUND-LIFT COMPROMISES SAFETY?
Based on the discussion above, ground-lift (that is not connecting signal GND to EARTH) does not seem to compromise safety. The fact that you can purchase double-isolated appliance with a two-wire power plug also says that it is not required to have a current path to EARTH in case of a fault. But don’t take my word for it. I am not a safety expert, just using a little bit of common sense. In my projects, I never connect the signal ground to earth ground but ALWAYS connect EARTH to the chassis and when connecting EARTH to chassis is not possible (like using a wood plate) then I make sure there is plenty of air gap between the component and my fingers…