The Soekris R-2R DAC: Technical Details

January 30, 2015 63 comments




The Soekris R2R DAC is finally out, officially named

The Soekris dam1021 Sign Magnitude R-2R DAC

I will be collecting useful information here and in other posts 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:

  1. I2S input up to 384KHz sample rate (tested to 192KHz)
  2. SPDIF input up to 192KHz (tested to 96KHz)
  3. Automatic De-emphasis for 44.1KHz material
  4. Built-in set of simple FIR filters for all sample rates
  5. Digital volume control through simple potentiometer
  6. Automatic input selection
  7. Data reclocking: s/w PLL with 0.02 Hz low pass filter
  8. S/W interface (serial interface) allows:
    1. Volume control (e.g. with Arduino and RS232 interface)
    2. Input selection (e.g. with Arduino and RS232 interface)
    3. Loadable FIR filters including bypass filter for NOS support (s/w utility included)
    4. Firmware update/upgrade

Upcoming features (through user-performed firmware upgrade)

Ref: [link], [link]

  1. DSD support
  2. Filter cascading for digital crossover
  3. Filter cascading for balanced use
  4. Master clock output
  5. 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
  6. Improved response for FIR2 filter which will be modified to be pretty soft….
  7. Enabled TTL-level serial port on the isolated side (3.3v)

Rather than incremental updates, Soren is planning a big update [link]



Ref: [link], [link]

The dam1021 DAC “raw” audio output is DC-coupled. The signal out of the resistor ladder passes through a low pass RC filter consisting of a 1200 pf ceramic NP0 type (“audio” grade), and the -3db point is at ~250 Khz (different values have been reported but at this cutoff, it doesn’t really matter to the audio frequencies).

The specs of the raw output are:

  • 0V offset (no need for DC blocking capacitor)
  • Output level: 1.4V RMS
  • Output impedance: 625 ohms, purely resistive.


The raw output is available for single-ended connection. It is also routed to a single-ended to balanced output buffer according to the following schematic:


The buffered output can be used single-ended or balanced. It can drive high impedance headphones directly.

The specifications for the buffered output are:

  • Output level: 2V RMS SE, 4V RMS BAL
  • Output Impedance: 10 ohm SE, 20 ohm BAL


Reference: [link], [link], [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 [901]

  • 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 [848]

  • 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 [1130]

  • 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.

1.2V Supply

The DC-DC regulators is the TI TPS562209 [link]

Here is the reference circuit taken from the datasheet:


3.3V Supply

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 need of the FPGA
  • Clean side of signal isolators
  • SPDIF LVDS receivers
  • Microprocessor
  • 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”.

Linear Regulator for +/- 5V

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.



Power Rails Connector/Header [1144]

J2 is a connector to all the power rails, for testing or for supplying (limited) power to other things. The 1.2V  can be used to bias the LVDS for Coax SPDIF input (see users guide)



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, 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? [1557]

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 format used in the FPGA of the dam1021 it’s 2.30 fixed point format for the FIR filters and 3.29 fixed point format for IIR filters. (2 bits for the integer part, 30 bit for the fractional part =32 bits). The mkrom utility (part of the filter tools) reads and process the input .txt parameter files as 64 bit floats, including the multiplier, then convert to the 2.30/3.29 fixed formats in the final step, which are pretty good. (The SigmaDSP chips use fixed 4.24 format for coefficients….)

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

Filter tools and documentation will be shortly available for users to upload custom filters to the FPGA.  The tools are already available here: [link]


The clock in the dam1021 DAC is the Si514. This the lower grade of programmable clocks from Silicon Labs (.8 psec RMS jitter) [link], and according to Soren, it is well matched to the system as a whole.  It is also used instead of the Si570 because of lower power consumption.


Master Clock Output Pin [848]

  • 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.


Here is a comparison jitter measurement between the Soekris DAC and other DACs [link], [link], [link]


Jitter reduction is accomplished with signal reclocking through a short FIFO. The data is received into a configurable FIFO and then it is reclocked with a lower jitter clock, eliminating most of the incoming jitter.

Further details [link]

The details of my clocking/FIFO:

Ian’s FIFO use a fixed clock, and therefore use a large buffer to take up the difference between incoming and outgoing clock. That add a large delay, which doesn’t matter for simple audio applications but are undesirable in a number of applications, like home theater or live music.

I use a much shorter FIFO, selectable down to 1 mS, and instead adjust the outgoing clock to match the incoming clock frequency as needed, being I2S or SPDIF. The Si514 oscillator used is very low jitter and digitally programmable with a resolution of 0.026 ppb (parts per billion, not million…). It also have the feature that reprogramming inside +-1000 ppm is glitchless, ie the clock adjust very nicely to small changes.


The DAC has two serial interface:

  • Standard RS232 serial interface
  • TTL level Isolated Serial interface (requires firmware > 0.9)

The RS232 interface is provided by the Intersil 3221ECVZ chip [link]

The Intersil ICL3221E devices are 3.0V to 5.5V powered RS-232 transmitters/receivers which meet ElA/TIA-232 and V.28/V.24 specifications. Additionally, they provide ±15kV ESD protection on transmitter outputs and receiver inputs (RS-232 pins).

Valid RS232 levels are >+2.7V and <-2.7V according to the datasheet. Strictly speaking is > +2.4V and < -2.4V but under +/- 2.7 V it may trigger the auto power-down.

Notice that pins 9 and 11 are the TTL level serial lines to/from the FPGA. The chip convert those signals to RS232 compatible levels and provide the robustness of the RS232 standard.




Automatic power down

The chip has  an automatic power-down function for noise reduction. When no valid RS-232 voltages are sensed on the receiver input for 30µs, the charge pump and transmitters power-down, thereby reducing supply current to 1µA.

The ICL32xxE powers back up whenever it detects a valid RS-232 voltage level on the receiver input. This automatic power-down feature provides additional system power savings without changes to the existing operating system.

The chip is a drop in Replacements for MAX3221E, MAX3222E, MAX3223E, MAX3232E, MAX3241E, MAX3243E, SP3243E, meaning it should be fully compatible with USB interfaces implemented with those chips.


Soekris dam 1021 R-2R DAC ILLUSTRATED GUIDE Users Manual [link] Users manual for the Soekris DAC.
Soekris dam1021 Build Build Guide [link] Details of my initial build of the Soekris DAC.
dam1021 R-2R DAC MODs Mods [link] Mods I have performed on the DAC build.
dam1021 R2R More Mods Mods [link] Later mods on the DAC build.
Digital Filters for Soekris R2R DAC Digital Filters [link] Extensive list of DIY filters from the diyaudio filter brewing forum thread.
R2R Benchmark Filters (for now) Digital Filters [link] Latest set of filters developed and shared in the diyaudio filter brewing forum thread. The best filters of the bunch.
R-2R DAC For The REST of US Technical Details [link] Introductory post describing the innovations and capabilities implemented in this DAC.
The Soekris R-2R DAC: Technical Details Technical Details [link] Additional technical details of the Soekris DAC that were not covered in the post above and collected after I had the DAC on my hands.


January 6, 2015 14 comments


Picked up a huge Marantz receiver (SR8200) at the local donation center for not very much money.


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]



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]


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


Output transistors: SANKEN A1492 (PNP) and C3856 (NPN)



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.



AKM Verita 4490EQ DAC

December 7, 2014 71 comments

(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):

Parameter AK4490
 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:

LPSRLinear 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.

LPSlRLinear 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).

MPSRMinimum 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

MPSlRMinimum 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

Parameter AK4490EQ  ES9018 ES9018K2M WM8741 PCM1794
DR (Mono) 123 dB 135 dB 127 dB 128 dB 132 dB
THD -112 dB -120 dB -120 dB -100 dB -108 dB
Max SR 768KHz 384KHz 384KHz 192KHz 192KHz
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:

I2S Pins
ESS9018 [link]
PCM1795 [link]
AK4399 [link]
Amanero [link]
SDTrans [link]
XMOS Ref [link]
BCLK DSD Clock DSD Clock DSD Clock DSD Clock DSD Clock DSD Clock
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):



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.


(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)

(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

(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)
(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)
(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)

(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
(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)

(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)

(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

Raspberry Pi B+ Digital Audio

November 13, 2014 16 comments


Although the Raspberry Pi has built-in analog audio output, the interest here is in digital audio output in particular I2S output signals for direct connection to digital to analog converters. I explored a bit the digital audio capabilities of the Raspberry Pi a while ago [link]. Here is an update with more accurate information.

The digital audio capabilities of the Raspberry Pi B+ have not changed from previous versions. The I2S audio is supported by the Broadcom BCM2835 [link]  peripheral SoC chip. The datasheet shows that the PCM audio interface consist of 4 signals, notice that there is no Master Clock signal:

  • PCM_CLK – bit clock.
  • PCM_FS – frame sync signal. Frames can be up to 32 bit wide
  • PCM_DIN  – serial data input.
  • PCM_DOUT – serial data output.

In addition, for more advanced configurations, the device can be configured as master or slave: “the direction of the PCM_CLK and PCM_FS signals can be individually selected, allowing the interface to act as a master or slave device”. In normal operation, it is configured as a master device.


In the Raspberry Pi B+, The I2S output are assigned to the following pins:



The audio frequencies (the PCM _MCLK) are supposedly generated by the use of a discrete 0n-board 19.2 MHz crystal. Unlike the BeagleBone Black, where there are facilities (pins) to feed an external master clock.


The frequency that is generated at any of the I/O pins, say the bit clock, is obtained by dividing the source clock (19.2 MHz oscillator) by configuring a clock division register with an integer part and a fractional part as indicated by the datasheet excerpt shown below:


The Datasheet indicates that the clocks are generated by “noise-shaping MASH dividers” which are fractional dividers. It also says that “The fractional dividers operates by periodically dropping source clock pulses”. I believe this post has an example on how this is actually implemented [link].

The way that a 3.25x clock divider is implemented is by dividing by 3x for some periods and 4x for other periods, with the average being 3.25x. In this case the repeating pattern will be (3, 3, 3, 4). That is shown in the following scope capture. Note that the first three periods are divided by 3 and then the next is divided by 4.
The way this is implemented in the device is to divide by the smaller divider and then extend the high pulse width by one clock cycle periodically.

We can find the integer divider and fractional divider based on MASH 1 (see above) and determine what is the maximum and minimum output frequency:

  • Source clock: 19,200,000 Hz
  • Sample rate: 44,100 Hz; bit clock (64fs)=2,822,400 Hz
  • Actual divisor: 6.8027. Integer part=6
  • Fractional divisor: =0.8027×1024=822 (round off)
  • Maximum frequency: 19,200,000/6=3,200,000 Hz (50 KHz sample rate)
  • Minimum frequency: 19,200,000/7=2,742,857 Hz (42.9 KHz sample rate)
  • Average frequency: 19,200,000/(6+(822/1024))=2,822,394 Hz (44.1 KHz sample rate)

Well, aiming at 44.1KHz sample rate frequency and getting a frequency variation from 42.9 KHz to 50KHz, this can’t really work for digital audio. Clearly there has to be a better way to generate these clock frequencies.


Much of the credit for enabling I2S output in the RPi (and the proper generation of clock frequencies) is due to the discussion in the Raspberry Pi forums [link] and work of Florian Meier in his master thesis “Low-Latency Audio over IP on Embedded Systems” [link] who subsequently developed the basic “ALSA SoC I2S Audio Layer for Broadcom BCM2708 SoC” audio kernel driver [link]

There it is explained that in order to get good clocks, one has to use integer division but with a 19.2 MHz clock, it is impossible to arrive at 32fs or 64fs bitclocks (e.g. 64x48KHz=3.072 MHz). Therefore other internal clocks must be used.

According to this post [link] the clocks sources are:

0     0 Hz     Ground
1     19.2 MHz oscillator
2     0 Hz     testdebug0
3     0 Hz     testdebug1
4     0 Hz     PLLA
5     1000 MHz PLLC (changes with overclock settings)
6     500 MHz  PLLD
7     216 MHz  HDMI auxiliary
8-15  0 Hz     Ground

The logical choices are the external 19.2MHz and the highest stable frequency clock which is the 500 MHz clock (highest frequency generates a more accurate clock after fractional clock division)

The author presents two solutions:

  • Use the 19.2 MHz oscillator with integer division for DACs that do not require a specific ratio of bit clock to frame sync (e.g. 32fs for 16 bit data) as long as there are at most enough bit clock cycles within a frame sync cycle to contain the desired word length
  • Use the 500 MHz internal PLL with fractional division for DACs that do require a specific ratio of bit clock to frame synch (e.g 32fs or 64fs)

The first solution says that it is possible to use, say 40fs, to sent 16 bit samples (16bitx2=32bit per frame) because you can transfer all 32 bits in a 40 bit frame. If you can use 40fs for the bitclock, then 40x48KHz= 1.920 MHz which is 19.2 MHz/10. The following excerpt from the thesis explains these two approaches:


We notice that integer division of the external 19.2 MHz clock only works for 48KHz and 96 KHz and for DACs that can operate at 40fs (80fs if we want to pass 32×2 bits per frame). The current version of the code is using 50fs and 100fs which also works.

For the 44.1KHz sample rate or for bitclock requiring 32fs or 64fs, then the first solution with fractional division is used on the 500 MHz PLLD clock


The fact that no clean clocks can be generated in the digital audio frequency range, tells us that this oscillator was not really meant to be used for digital audio. Now why did the designers of RPi use a 19.2 MHz clock?

I have searched extensively and cannot find a “reason” for the 19.2 MHz frequency. If it were digital audio, a more logical selection would have been 24.576 MHz in order to cleanly support 64fs 48KHz sample rate (like the BeagleBone Black).

A better reason is to use this clock for the on-board PWM audio. One can easily generate a 48KHz carrier frequency (19.2MHz/40) and a resolution of 16 bit would require a frequency of approx 64 KHz (19.2MHz/30). In actually, it has been reported that the resolution is in the neighborhood of 11 bit or 2048 levels which can be obtained by dividing 19.2MHz by a factor of 9375.


A better solution is to configure RPi as a slave device and the DAC as a master device.

The DAC can provides a much more accurate clock to the RPi by feeding the Bitclock. I don’t think is being done by the current crop of DACs (the ones based on the PCM5122)  but the capability is there for both in the RPi (“clock slave mode” and “frame synch slave mode”) and the PCM5122 as shown in the following excerpt from the datasheet:


Here is what is required to set the DAC in Master Mode, say for example the PCM5122.

  1. RPi detects the sample rate of the clicked-to-play track.
  2. RPi has a way to indicate the sample rate (for example using GPIO pins).
  3. Microcontroller reads sample rate.
  4. Microcontroller programs the appropriate frequency by generating an appropriate master frequency from the PLL and setting the appropriate divider to generate the bitclock.

Other considerations:

  • Timing of the different events. For example, wait until the microcontroller programs the DAC to the appropriate clock frequency before staring the data stream (DMA) in the RPi.
  • Selection of external clock. For example use a single frequency clock, say a multiple of 44.1KHz in order to take advantage of integer divider only when dealing with frequencies multiple fo 44.1KHz.


Here is the previous comparison table with some updated observations (italics)

Parameter Rasberry Pi
BeagleBone Black
Native I2S support Yes Yes Both platforms can support I2S output, Custom drivers have been developed by the audio community
I2S Sample Rate limitation Up to 192KHz (because the on-board clock is 19.2MHz) Only 48KHz family (because the on-board clock is 24.576MHz and integer clock dividers) BBB supports 48KHz, 96KHz, 192KHz and 384KHz. RPi supports 44.1KHz, 48KHz, 88.2KHz, 96KHz, 176.4KHz and 192KHz (in theory). RPi uses “fractional clock dividers” to generate the 44.1KHz sample rate family as explained above
Support for USB DAC Yes (LAN9512 chip [link]) Yes (Built-in in the main processor) USB in the RPi goes through a built-in HUB and it is shared with the LAN controller within the USB/LAN chip. USB in the BBB is natively supported by the main processor; LAN has a separate chip
Support for external, low jitter clocks Not possible unless you replace the on board oscillator and modify the driver Yes with custom boards and custom s/w: [link] The master clock in BBB may be provided externally by disabling the on-board audio-freq clock.The Master clock in the RPi seems internally generated and there is no I/O pin to feed an external master clock
Master clock output No Yes (from on-board clock) The Master clock in BBB is provided by the on-board 24.576MHz and fixed at this frequency and can be directly accessed from the outside. The Master clock of RPi seems internally generated but un-accessible from the outside. Without Master Clock, you can only use DACs that can operate asynchronously without a Master Clock input such as the ESS DACs or DACs that can operate with the master clock = bit clock
Built-in rechargeable battery operation No Yes [link]. Maybe Rechargeable Battery operation in BBB would disable the 5V supply to the USB. Thus for USB operation, where the USB adapter takes the power from USB, BBB must be powered with 5V DC
Built-in Storage No.  But the new model has plenty of USB ports for USB memory sticks 2 GB eMMC Flash BBB can boot from the internal storage freeing the SD card for music storage. RPi requires that the OS be stored in the SD card (although it may be possible to also store music in the SD card)
Looks The latest model looks Good Good 🙂
Audio H/W and S/W community support Large Small
Price $35 $55

Here is a summary of the phase noise measurements from this post [link]:

I2SPhaseNoise 2


  • The ESS9023 implements a “jitter eliminator” (asynchronous sample rate converter) but cannot eliminate all the jitter
  • The clocks on the embedded boards have a lot of jitter. It also makes sense that the BBB has better measurement than the RPi. In the BBB, the 48KHz sample rate frequency is derived by integer division of the external clock. In the RPi, the 44.1KHz sample rate frequency is derived from the 500MHz clock which is derived from the external clock as explained above
  • The “lowly” CD player is still a pretty good playback device in terms of jitter
  • I would guess (and only a guess since the author does not identify these interfaces) USB-I2S-2 is an XMOS-based device based on how the clocks are generated and that USB-I2S-1 is a device based on an FPGA or CPLD using two external audio frequency clocks (where straight integer division is used)


The current method of generating the clocks for digital audio in the RPi are far from perfect. The best clocks are obtained by integer division of the external clock and works for 48K and 96K sample rates and only if the DAC can accept 40fs or 80fs. For anything else, the clocks are derived from the 500MHz PLL through fractional division as explained above. It has been reported that the 500MHz clock itself is derived from the 19.2MHz external clock through a clock multiplier.

However imperfect as these clocks might be, there are a good number of DAC boards that have been developed and reported to work well with the RP1. As these products are being developed by audio fanatics, we can expect continuous improvements and enhanced approaches to better clock generation such as external reclocking and slave configurations.

For additional info, you can check the Raspberry Pi I2S discussion thread here: [link]

Raspberry Pi version B+

November 12, 2014 15 comments

Previously I wrote:

I had ordered a BBB for no other reason that it’s better looking than the Rpi 🙂

Well, no longer. I ordered the new version B+ because it is as good looking as the BeagleBone Black 🙂



Not really. The reason is because that there is a lot, lot more community development in the Pi than the Beagle. In fact if we just look at shipments, the Raspberry Pi sells almost 20 times the amount of BeableBone Black (over 3.8 million [link] vs 200,000 devices [link]). This is a huge advantage in the popularity front.

I had been a fan of the BeagleBone board [link] mainly because it is a higher performance board and had local storage. In addition (by design or by accident) the audio master clock uses an on-board 24.576 MHz clock from which it derives the frequencies for 48KHz sample rate material with integer division. There is also the capability of receiving the master clock from external sources and thus it is possible to feed it a higher quality 24.576 MHz and 22.5792 MHz clocks. There has been a clock board (and corresponding drivers) in the works since early this year, but nothing available yet as of this writing [link]

In contrast, there has been a much larger development effort in the RPi front as testified by the numerous I2S DAC boards that have become commercially available. Many of these companies are dedicated to building audio solutions first for the Raspberry Pi and then possible for other embedded platforms.

Here is a list of DAC boards available for the Raspberry Pi (versus none for the BeagleBone Black as of this writing)

G2 Labs BerryNOS 1543 RED $125 Philips TDA1543 Balanced design, discrete output stage, power supply
BerryNOS mini $60 Philips TDA1543 Balanced design, discrete output stage
HIFIBerry HiFiBerry DAC €25 TI PCM5102A
HiFiBerry DAC+ €30 TI PCM5122 DAC volume control
IQAudio PiDAC $38 TI PCM5122 DAC volume control
PiDAC+ $42 TI PCM5122 DAC volume control, headphone amp
Saparel RaspiPlay3 $40 TI PCM5102A From Serbia
RaspiPlay4 [link] TBD TI PCM5122 DAC volume control, IR remote
Audiophonics I-Sabre DAC €25 ESS ES9023
I-Sabre DAC+ €43 ESS ES9023
Element 14 Wolfson Audio Card $35 Wolfson WM5102 Available through resellers. WM5102 is a complete audio system. The board implements line-in, line-out, speaker and headphone output and mic input. The board also includes a WM8804 providing SPDIF input and output, a digital microphone and expansion header for other Wolfson devices
TekDevice DACBerry2+ $45 TI PCM5102A
DACBerry3+ $51 ESS ES9023
HIFImeDIY ES9023 DAC $19 ESS ES9023 Lowest price!
DurioSound DurioSound $45 TI PCM5102A Has ultra low noise regulator (TPS7a47)
DurioSound Pro $70 TI PCM5102A Has ultra low noise regulator (TPS7a47) and Local Power Supply


  1. Notice that the DAC chips used are the ones that can cope without a Master Clock. RPi I2S does not Master Clock, so the DACs synch on bitclock and generate their own master clock.
  2. Products using PCM 5122 can use the DAC’s internal volume control and therefore can be connected directly to an amplifier.
  3. There are companies such as diyinhk and curryman that are not listed because they do not specifically make DAC boards that conform to the RPi footprint but are fully functional as I2S DACs. Any I2S DAC that does not require master clock will work.

My Favorite ones are the HIFIBerry DAC+ and the IQAudio PiDAC+, both based on the PCM 5122 with “hardware” volume control (meaning using the volume control in the DAC itself)






In the two years since we launched the current Raspberry Pi Model B, we’ve often talked about our intention to do one more hardware revision to incorporate the numerous small improvements people have been asking for. This isn’t a “Raspberry Pi 2″, but rather the final evolution of the original Raspberry Pi. Today, I’m very pleased to be able to announce the immediate availability, at $35 – it’s still the same price, of what we’re calling the Raspberry Pi Model B+. [link]

There are a million reviews on the Raspberry Pi. Here is one more but with a slant towards diyaudio…

New Layout (and more I/O pins)



Notice that the I2S pins are right next to a GND pin. This is particularly good as you can easily use twisted pairs (for noise immunity) when connecting to a DAC

List of integrated circuits [link]

Label Device Description
U1 BCM2835 SoC comprising ARM Processing core and Video Core. Data Sheet
U2 LAN9514 4 USB 2.0 Hub and 10/100 Ethernet controller. Data Sheet
U3 PAM2306AYPKE Dual DC-DC Switching converter. Data Sheet
U4 APX803-46SAG Brownout detector (reset generator)  Data Sheet
U5 AP7115-25SEG 150 mA Linear Regulator. 50 uV noise (Video DAC)  Data Sheet
U6 N.U.
U7 N.U.
U8 ESD5384 ESD protection for HDMI. Data Sheet
U9 AP2331W Current limited switch (for HDMI hot plug) Data Sheet
U10 AP7115-25SEG 150 mA Linear Regulator. 50 uV noise. (PWM Audio Driver supply) Data Sheet
U11 NC7WZ16 Ultra High Speed dual buffer. (PWM Driver) Data Sheet
U12 N.U.
U13 AP2553W6 USB current limited power switch (for hot plug). Data Sheet
U14 DMMT5401 Matched PNP transistors. Data Sheet

Board schematic here: [link]

The audio jack is also a composite AV jack


More USB Ports


New USB/Network Chip (to support the 4 USB ports)


The USB powerchain has a proper limiting switch and will not brown out the board if USB devices are plugged in when powered (or even if they try to take too much current or there is a fault like a power short). Default allowed USB current across 4 ports is 600mA, but can be increased to 1.2A via a config.txt parameter if a good quality 2A PSU is used. I have tried a few different USB hard disks and they all power fine directly from the Pi at the 1.2A setting. [link]

To increase power to 1.2A you add the following line in /boot/config.txt[link]

  • max_usb_current=1 (newer software)
  • safe_mode_gpio=4 (older software)

The 5V for the USB ports is provided directly by the 5V of the input supply. The schematic below shows the 5V sourced from Power In. There is a 2A fuse a diode-like low-drop polarity protection circuit and an over-voltage zener.


Therefore a better quality power supply is required. Here is an excellent post on choosing and evaluating 5V charger/supply [link]

I like the Orico DCX-2U. It has two USB outputs: 1×2.4A, 1×1.5A. The 2.4A output is plenty for a “fully loaded” RPi

High Efficiency Switching Supply (power consumption is reduced by between 0.5W and 1W)


The DC-DC Switching supply is the PAM2306AYPKE. This device supplies the 3.3v and 1.8v supplies. The switching frequency is 1.5MHz (which can easily removed by the LC filters on the outputs).


MicroSD Card Slot

There are many theories as to why the SD card was replaced with the micro SD card. I think it is probably lower cost.



According to the people from Raspberry Pi, the audio in the B+ model has been improved. The audio circuit (AUDIO out) incorporates a dedicated low-noise power supply (the input to this power supply is the external 5V supply). According to this comment [link]:

The B+ does not use use a switching regulator for its PWM driver, that would indeed be a bad design choice, instead it uses the AP7115-25SEG [link] low drop regulator with high power supply rejection ratio. It creates a noise free 2.5V for the NC7WZ16 PWM driver, the output of which is attenuated and filtered with two 100 Ohm resistors, and a 100 nF capacitor, so the output is 50 Ohm, and can reach 1.25Volt p/p.

U10 is the linear regulator and U11 is the “PWM Driver”


Whether this audio is “better” or not, it does not concern us. Take a look at this post [link].


Same SoC (Same ARM processor and GPU and 512MB of RAM)


Unlike the BeagleBone Black, there is no local memory for storage. The s/w runs out of the microSD card.

Same external 19.2 MHz crystal


First One Build: Adjusting Operating Parameters

November 11, 2014 10 comments

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




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]


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)


First One Build: Ground Connection

November 6, 2014 7 comments

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.

(In fact…)

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.


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.


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…

First One Build: Power Supply Selection

October 31, 2014 8 comments

LC tested three First One amplifiers, each one equipped with different power supply [link]:

  • Hypex SMPS1200A400 [link]
  • AudioPower DPS-500/63
  • Connex SMPS2000R [link]

AB comparison 1: Single Connex vs Dual AudioPower


DPS500 2-001

It was not really hard to recognize the better PS, since bass was really on a weak side of Connex, probably the reason lies in the use of a single SMPS2000R for both amp’s channels, nevertheless the sound-stage was narrow, instruments weakly presented, thin bodies, no real feeling of involvement into the music. DPS-500/63 were in dual mono configuration, meaning one per channel. Transparency and details resolution was very similar but the bass was one step ahead over the Connex. Still no real musical involvement presentation here, sound more or less stuck to the speakers, instruments with weak bodies, sound-stage not well formed.

Since both were regulated SMPSs we got an impression that the problem lies in the voltage regulation principle, too much interfering in audio signal is probably the worst thing it could happen from amp’s power supply.

It is interesting to point out that without AB comparison, LC had this to say about the Connex supply:

Cristi, your second SMPS2000R is just WOOOW, driving both channels. This is the one packed with Rubycons on primary and secondary side, set to +/-63 V, no hum, dead quiet silence, rock stable imaging, resolution is at the top notch, no sign of sibilants on vocals, liquid like sound, bass is simply awesome, like a day and night different from first one tried, also much better than SMPS1200 [link]

Yes, a day and night between the two SMPS2000R, named them A and B versions. As I can see the only difference is in secondary cap bank, maybe also something else, surely Cristi would have something more to say about them. Anyway version B with Rubycons is an absolute winner up to now, doubtless.

SMPS1200 provided better bass than Cristi’s A version, but that’s about it, both Connex gives higher headroom, also more power because of higher and regulated rail’s potential, at the end resulting in more stable sound imaging. [link]

AB comparison 2: Single Hypex vs Dual AudioPower


Here is the latest version of the Hypex supply. The entire PCB was updated in 2013 according to the data sheet [link]


After half an hour of mental pause, test continued between (single) Hypex SMPS1200A400 and (dual) Audio Power DSP-500/63. First we listened to DPS-500/63, musical impression stayed very much the same as it was in the first session. Weak presentation left us more or less cold, music is simply too thin, uninvolving.

Hypex’s turn then settled the things where they should belong, suddenly real music in the room. It was immediately clear to all of us who’s having magic stick in its sleeve. SMPS1200A400 shocked us with music so real that the other two seemed like that there was something broken in them.

Hypex presented bass so low and strong that our jaws just dropped for a while, dense wall of air moved with lowest possible frequencies we could still hear, still with ease. Sound-stage completely another story from previous contenders, no speakers in the room only musicians and instruments, this time with fully developed bodies, atmosphere of a recording stage, whether real or artificial, presented in full scale. SMPS1200A400 puts the First One amplifier in the first league of power amplifiers no matter the price level.

AB comparison 3: Single vs Dual SMPS1200A400 [link]


The result is not so far from our expectations, tight similarity with very slight differences noticeable on momentary A-B test; otherwise, on not so closely conducted comparisons, it would be very hard or even impossible to distinguish the two configurations.

In which cases to choose single or double SMPS solution greatly depends on the speaker’s efficiency and impedance, these two parameters dictate how power hungry your speakers are and of course how loud you want them to be.

The ultimate and preferred First One solution is still dual mono configuration, although the use of a single SMPS completely fulfills the needs in a smaller system.

AB comparison 4: Linear vs AudioPower [link] (Comparison on-going…)



Well, after the not-so-subtle remarks in the AB comparison performed by LC, it is almost impossible to argue the fact that using anything else than a Hypex SMPS1200A400 supply would “rob” performance out of the amplifier modules. A plus is that the amp modules are factory calibrated and tested with this supply in mind. In addition, especially for a budget-constrained build, the AB comparisons also showed that unless there is a need for high current demand, a single SMPS1200A400 would sufficiently fulfill the designed performance of the First One amp. Further, the Hypex SMPS1200A400 is competitively priced against the other two offerings.

Best choice

If one wishes to confirm the results or adjust certain variables in the comparisons, it would be difficult for the diyer to replicate even some of the tests reported by LC. First several amps were configured with the different supplies for a immediate AB comparison. Second, one would have to procure the different supplies for the test. This would be cost prohibitive for the common diyer. In addition Hypex has a long and excellent reputation for high end audio not just a a provider of high fidelity products but also as a technical innovator, and thus there is very little chance to find a superior supply for this application.

Why Hypex supply outperforms Connex supply?

Having examined the Connex supplies in detail, I find that the use of soft switching approach minimizes the generation of EMI while increasing efficiency. The hypex datasheet does not say whether it is a “soft” or “hard” switching approach. It merely says “The SMPS1200 is optimized from the first phase of design to final implementation to realize the lowest possible EMI signature required of the most demanding audio applications” this could very well mean the use of soft-switching and/or more aggressive output filtering. So from a SMPS switching approach point of view, one cannot say why one would sound better than the other.

Therefore, the difference in audio quality seems to be fully attributed to the unregulated output nature of the Hypex supply. This comes as a surprise because the audio implementations have been moving toward regulation and now we find that SMPS with regulated output seems detrimental to audio quality at least in this instance.

LC believes that regulation “interferes” with the audio signal [link]

For those interested in more technical details, the Connex supplies were tested extensively [link]. And even thought they exhibit superb performance and I remain a fan of Connex supplies [link], I cannot justify using them with the First One Amp modules in light of the comparison presented here.

AudioPower develops Unregulated PS

It is worthwhile to note that AudioPower has recently developed Unregulated versions of their Audio SMPS, perhaps a testament that “unregulation” has sonic advantages (or just competing with Hypex). [link]. I’ll have to admit, they are best looking.


My initial choice

For now I will use the power supply of the Adcom GFA-5300 AM. It generates +/- 52 volts and according to spec, can supply a max of 720VA


First One Build: Heatsink Selection

October 31, 2014 3 comments

One of the most important parameters for proper operation of the Fist One amplifier module is adequate heat dissipation through a large enough heatsink. The amp module operates in class AB with an idle current of 280 mA.

The idle dissipation of the First One is >30 W at +/-63 V, plus audio power dissipation easily adding extra 50-70 W, so 100 W all together to dissipate.

To calculate the temperature of the heatsink during operation: 100 W * 0.5 K/W=50 K added to room temperature (25 C) resulting in 75 C heatsink temperature. At that point silicon die in output transistor is around 100 C and that is somehow at max acceptability. To calculate idle temperature: 30W* 0,5 K/W=15 K (or C) added to room temperature resulting in 40 C. [link]

Supply DC current to the First One module without input signal present (idle current) is 280 mA, multiplying it with 120 V rails potential, gives 33.6 W of total quiescent power dissipation per module, so in stereo total 67,2 W. That is serious thermal loading for the chassis and heatsink if one would want amp to be below 45 degrees in a room environment. [link]

LC recommends “any heatsink having 0,5 K/W or even lower”. Something like Fischer Elektronik FK157 [link]. Below are the heatsink profiles of the FK157 and other similar profiles that will yield 0.5 K/W dissipation or better. These were extracted from the Fischer Elektronik catalog [link]


Notice that by comparing the 3 profiles shown above, in order to achieve a 0.5 K/W dissipation you would need:

  • 2″ of SK501
  • 2″ of SK 586
  • 2″ of SK 157

Seems longer fins only help if you need a dissipation factor lower than 0.5 K/W or even lower than 0.3K/W

Even a shorter profile would yield a dissipation rate of 0.5 K/W. In this example a 4″ heatsink would achieve a dissipation rate of 0.5K/W


The Semelab application note has a extensive section on heatsink selection [link]. If you read the whole thing, basically the bottom-line thing to do is that the heatsink shall not exceed 70C during operation.

Factory Chassis

LC provides appropriate heatsinks as part of the factory chassis (which cost Euro 300 plus shipping). Following are the photos of the “factory” heatsinks ( I think they are SK 157 with height 70 mm, so having a dissipation coefficient of a bit less than 0.4K/W). The chassis is beautiful and built like a tank. If you want the best, this is it.




In order to obtain the factory heatsinks, you need to purchase the chassis (300 euros plus shipping – I would think US$60-$100 for shipping based on eBay examples. So total cost would be US$450-$500) [link]:


Although it is highly desirable to have an enclosure that is built at the same high standards as the amp module, if budget does not permit, there are other options.

Chinese chassis from eBay

The is the the lowest cost for a chassis plus heatsink meeting the required dissipation rate [link]. This case costs about US$ 160 including shipping.


The heatsink size for this case is 300x50x67mm with a profile similar to SK501 but with the fins 10 mm longer. At 67 mm height, Likely it exceeds the required 0.5 K/W dissipation rate. It probably rates at 0.45K/W. (This is just theory in practice you may need a larger (taller) heatsink depending on different factors such as ambient temp, etc)

You can find an example implementation of this case with the VSSA amp here [link].

A similar but with taller heatsinks can be found here [link] and here [link]

Heatsink Only

If you want the minimum cost and If you live in the USA, a good source is “HeatsinkUSA”. High quality and good prices. The largest one seems to fit the bill [link]. Specs are:

  • Width is 10.080″
  • Fin Height is 2.5″
  • Base Height is .375″
  • Weight is approximately .99 lbs per inch
  • C/W/3″: approximately .80 (for a 3″ heatsink)


This heatsink is similar in profile to  SK524 above except it has one less fin but the fins are much larger at 50 mm. If we use the dissipation curve of the SK524 we find that a 4″ heatsink will meet the required 0.5K/W dissipation. Note that the published thermal dissipation specification for this heatsink is 0.8 s for a 3″. If we go by the dissipation curves shown in the Fischer Elektronik catalog, then this values seems too conservative. But in order to be safe, a 5″ heatsink would likely be more than sufficient.  A pair of 5″ heatsinks would set you back about $90 including shipping.

Thrift Store Amp

Even cheaper than getting heatsink is using an old amp from a thrift store. If you are lucky, you may find an old amp with large heatsinks. I had purchased a used Adcom GFA-5300 amplifier from the local thrift store for $15. This was a few years back. nowadays, even thrift stores are drastically increasing the price of used audio equipment. I would say this amp would probably sell for $50 if bought today.

The heatsink of the Adcom has the following dimensions:

Width: 200 mm; height: 90 mm; depth (fins): 55 mm; base plate thickness: 5 mm; number of fins: 20.



The closest profile I could find from the Fischer Elektronik catalog is the following:


As can be seen, the Adcom heatsink is a bit wider, the fins a tad longer and it has 4 more fins. I would say at 90 mm in height, it would easily meet 0.6K/W. but it does not meet the minimum requirement of 0.5K/W.

Using the power supply of the Adcom Amplifier

What if we use the power supply of the Adcom which provides +/- 52V? We can calculate the required heatsink dissipation with this supply by following the example given at the beginning of this section and the following requirement [link]:

As we don’t want to have more than 45 C in idle, please use heatsink having thermal coeficient of 0,5K/W or less for each channel.

First One module has 35 W idle power dissipation when supplied from +/-63 V PSU.

The idle current of the Amp is 280 mA (how to measure [link]). Even at a lower supply voltage this bias requirement is fixed [link]

Thus at +/- 52V supply we get 29 Watts. With a heastsink of 0.6K/W we get 29*0.6=17.4 C. Adding the room temperature of 25C we get 42.4K which is within spec but this is only at idle.

The service manual of the Adcom Amp gives this power data:


I plan to use this Adcom amp for my first build. It seems to have adequate heatsinks. I will have to build up the amp to know for sure.

Summary of choices

  • Factory case: ~$450-$500
  • eBay Chinese case: ~$150-$200
  • Heatsink Only: ~$90
  • Thrift Store Amp with large heatsinks: ~$30-$60

First One Amplifier Module

October 31, 2014 14 comments

The First One Amp module [link] is a high performance (High Fidelity?) and yet very affordable class AB current-feedback amplifier module.  It establishes a benchmark for price/performance.

Developed by “Lazy Cat” (LC) at diyaudio, it is the big-brother commercial version of the DIY VSSA (“Very Simple Symmetric Amplifier) [link] and incorporates all the knowledge obtained from that project. Whereas the VSSA was fully open and fully diy, the First One amplifier is available as a factory built and tested module. Available for diyers as well as OEM to manufacturers, the module has been seen in a finished amplifier for a road show in Slovenia [link].

Photo of First One’s little brother: completed VSSA module (built by LC):



Since I am new to this module and was not aware of the VSSA, I’ll use this and following post to gather my knowledge for my amplifier build. The information is mostly from the diyaudio threads, but there it  is spread out all over the place and hard to find.

Photos of the First One Amplifier Module.


Use of name-brand “audio grade” components…


The current version is V1.2. There is a V1.3 that has been developed but not quite yet available for sale. For those of us with V1.2, LC has promised to send modding instructions but only to those that have completed the build of the amp.


Thermal coupling for these two transistors. The schematic is not public since this is a commercial product.


Notice the adjustment pots (TRx) and the measuring points (TPx)


Output Power transistors are Semelab “ALFET” double die MOSFET N and P-channel pair, rated at 250 W and 16 Amp continuous current. These are specially designed for audio applications [link]:

  • The N-channel device is: ALF16N16W/ALF16N20W [link]
  • The P-channel device is: ALF16P16W/ALF16P20W [link]




Module Size

100 x 50 x 40 mm (W x D x H). [link]

Supply Voltage

+/-40 V to +/-63 V

DC coupled

The amp modules are DC coupled, no capacitor in front of the input stage.


I’ll compare the specifications of the First One module (FO) [link] vs an old Hitachi amp [link] and an Adcom amp:

Parameter First One
Hitachi HMA-7500
Adcom GFA-5300
Max Power 8 Ohm 150 Watt 0.05 THD 80 Watt 0.005 THD 80 Watt 0.018 THD
Max Power 4 Ohm 230 Watt 0.05 THD 80 Watt 0.005 THD 125 Watt 0.018 THD
Bandwidth 3 Hz to 3 MHz (-3dB) 5 Hz to 100 KHz (-1dB) 3 Hz to 130KHz (-3dB)
THD 0.0034% (100 Watt) <0.005% (80 Watt) 0.02% (125 Watt, 1KHz)
IMD 0.003% <0.008% <0.07%
SNR 110 dB 118 dB >100 dB
Input Impedance 10 Kohm 47 Kohm 50 Kohm
Damping Factor >2000 (4 ohm) 100 (8 ohm, 1KHz) >350
Year Introduction 2014 1980 1995

According to published specifications, the First One amp has very impressive specifications and overall best of the bunch. The old Hitachi has still has very impressive specifications (but at a much lower max power).

Damping factor

Measurements performed in order to determine Zout and consequently the damping factor (DF). A sinusoidal signal of 100 Wrms at 20 Hz, 1 kHz and 20 kHz was passed onto a 4.08 Ohm load resistor, measured with FLUKE 289 True RMS Multimeter and here are the results. [link]

20 Hz, 100 Wrms/4.08 Ohm:

  • DF(20 Hz)=Rload/Zout=4.08 Ohm/0.00121 Ohm=3372

1 kHz, 100 Wrms/4.08 Ohm:

  • DF(1 kHz)=Rload/Zout=4,08 Ohm/0,0004 Ohm=10200

20 kHz, 100 Wrms/4,08 Ohm:

  • DF(20 kHz)=Rload/Zout=4,08 Ohm/0,00162 Ohm=2519

Very large damping factor by itself likely means that the amp itself would not be the limiting factor for controlling the oscillations in the speaker. This means that other factors (such as speaker cable impedance) would contribute more to the damping factor seen by the speaker. The speaker’s own impedance is the mayor contributor…


Seems a perfect match for the upcoming discrete R2R DAC. The amp being single-ended (and DC-coupled) can take the output signal straight out of the resistor ladder. In addition, being wide-band would further benefit from R2R conversion (as opposed to delta-sigma) because the R2R DAC does not generate high frequency noise.