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…
LC tested three First One amplifiers, each one equipped with different power supply [link]:
AB comparison 1: Single Connex vs Dual AudioPower
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.
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
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.
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].
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.
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
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):
FIRST ONE MODULE
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]
100 x 50 x 40 mm (W x D x H). [link]
+/-40 V to +/-63 V
The amp modules are DC coupled, no capacitor in front of the input stage.
|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)|
|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|
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).
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…
A PERFECT MATCH WITH R2R DAC?
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.
A MOST INTERESTING DIY PROJECT IN A LONG TIME
A Discrete R-2R Sign Magnitude 24 bit 384 Khz DAC [link].
The DAC Module includes all local power supplies, a programmable low jitter clock, Micro-controller and balanced output buffer. It is implemented on a 4-layer PCB. The board size is 3.2″ x 5.8″ (81 x 147 mm).
As the industry migrated from R2R topologies to Sigma-Delta in their quest for higher bit-depth, higher performance (and cost management), present implementations of R2R DACs are pretty much hand-crafted commanding a high premium.
As the author states:
“I believe that the sound quality will be the absolute best, better than any Delta Sigma DAC, in class with discrete DAC’s from totaldac and msb technology. And for way way less cost :-)”
For the rest of us with limited resources wanting to experience a ladder DAC, this is the DAC to have.
An excerpt from the PCM1704 [link] datasheet expunds the good points of a ladder DAC:
Digital audio systems have traditionally used laser-trimmed, current-source DACs in order to achieve sufficient accuracy.
However, even the best of these suffer from potential low-level non-linearity due to errors in the major carry bipolar zero transition. Current systems have turned to oversampling data converters, such as the popular delta-sigma architectures, to correct the linearity problems. This is done, however, at the expense of signal-to-noise performance, and the noise shaping techniques utilized by these converters creates a considerable amount of out-of-band noise. If the outputs are not properly filtered, dynamic performance of the overall system will be adversely effected.
The PCM1704 employs an innovative architecture which combines the advantages of traditional DACs (e.g., excellent full-scale performance, high signal-to-noise ratio, and ease of use) with superior low-level performance.
Granted, that was circa 1999. Since then the Sigma-Delta camp has made great strides. Even so, R2R DACs have not lost their appeal as witnessed by the interest in this project and the current commercial offerings.
The DAC module is not yet available for sale. The target price is US$240 with 0.02% resistors.
ADVANCING THE STATE OF THE ART
The last commercially available R2R DAC chips were the PCM1704 [link] and the AD1865 [link]. They have been out of production for a long time but still available for purchase for example here [link] and here [link].
Here is a table comparing selected performance numbers and features as described in the data sheets and by the author in the diyaudio discussion thread.
- The PCM1704 is typically used withe a companion chip, the DF1704 [link].
- The AD1865 is also used with a companion filter chip such as the Sony CXD1244S [cxd1244s]
|Max Input Sample Size||24bit||24bit||18bit|
|Max Input Sample Rate||382KHz||96KHz||44KHz|
|Max Resolution||28bit (1)||24bit||18bit|
|Inputs (2)||1x Isolated I2S, 3x SPDIF/TOSLINK/AES/EBU [link]; future DSD upgrade||Serial only (DF1704: LJ, I2S)||Serial only through the digital filter chip|
|S/W Interface||Serial (Not I2C)||Serial (Not I2C)||Depends on filter chip|
|Oversampling Filter||On-board built-in and user defined (3)||Sharp, Slow roll-off (DF1704)||Needs External Filter|
|Channels||2 – Stereo||PCM1704 is single channel, so DF1704+2XPCM1704||2 – Stereo|
|Jitter Reduction||Re-clocking input data through a FIFO Buffer (similar in design to Ian’s FIFO [link]). Uses a low jitter (0.8 psec RMS) Si514 programmable clock [link] which drives the LVC595 shift registers after clock division in the FPGA (Si514 -> FPGA divider -> LVC595)||None||None|
|Output||“Raw” single-ended voltage output (1.4V RMS, 1.25 Kohm) or buffered balanced voltage output using TI LME49710 + LME49724 [link]||Single-ended current output||Single-ended current output or buffered single-ended voltage output|
|Jitter Reduction||FIFO Buffer and reclock with low jitter clock||None||None|
|THD+N (0db)||0.0063% .05% resistors (Module measurement)||0.0008% K-Grade (PCM1704 spec)||0.003% J/K Grade (AD1865 spec)|
|THD+N (-20db)||-||0.006% K-Grade (PCM1704 spec)||0.01% J/K Grade (AD1865 spec)|
|THD+N (-60db)||0.37% .05% resistors (Module measurement)||-||1% J/K Grade (AD1865 spec)|
|SNR||126 dB (Link)||120 dB||110 dB|
(1) The Soekris R2R implements 28 bits of internal resolution in order to provide sufficient headroom to allow for a “perfect digital volume control. At -72 db volume you still have 16 bit resolution with perfect linearity” [link].
(2) The PCM1704 and AD1865 are NOS ladder DACs expecting an input stream from an external filter device such as the DF1704 [link]. Therefore they typically cannot accept and I2S input format. The input format for those chips consists of a clock signal, data signal and data latch signal. More information can be found in Ian’s “I2S to PCM” board project [link].
(3) The oversampling filter is implemented in the on-board Spartan-6 LX16 FPGA. It has 15K logic cells and can be configured as having 8 full high resolution MAC’s by using its 32 DSP48A1 MAC blocks in groups of 4 allowing them to do 35 x 35 bit multiplications plus 70 bit summers. Two of these hig-res MACs can be used for the first 2 most critical oversampling FIR filters; running them at just 49.152 Mhz makes space for 1024 coefficients if needed, then 2 more for the rest of the FIR filters. The rest for other functions, like de-emphasis, volume control and digital crossover filters… [link]. The user can use generate the filter coefficients and upload them to the FPGA [link]
Here is a photo showing some of the details disclosed in the diyaudio thread
The input is isolated with (what appears to be) TI ISO7420FE digital galvanic isolators [link]. There are 3 identical isolators resulting in 6 input lines. I think these support one I2S input and 3x SPDIF/TOSLINK/AES/EBU (I don’t know if the SPDIF lines are isolated, but there is no need for 6 isolated inputs if only the I2S is isolated). More info on isolators here [link]. Seems everyone has their favorite isolation device. Of the 4 different vendors I have surveyed, they have all been used by different audio diy implementers.
The TI ISO7420x and ISO7421x provide galvanic isolation up to 2500 V RMS for 1 minute per UL and 4242 V PK per VDE. These devices have two isolated channels. Each channel has a logic input and output buffer separated by a silicon dioxide (SiO 2 ) insulation barrier.
Built-in galvanic isolation at the input is a great idea. This gives the capability to completely isolate noise disturbance is coming from the source, including isolating ground, and since here is a FIFO reclocking stage afterwards, there is no need to worry about the small added jitter (100-200 psec RMS) that these devices would add to the data.
A notable feature of this DAC module is the reclocking of the incoming. The design is similar in principle to Ian’s FIFO reclocker, The data is received into a configurable FIFO and then it is reclocked with a lower jitter clock.
However, Ian’s reclocker is designed for ultimate performance, whereas this reclocker is designed specifically for the DAC module and therefore matched to the requirements of the entire system (meaning, I think, the best consideration for jitter performance, cost and part count).
Here are the main differences between the two:
Ian’s FIFO reclocker
- Clock is Si570 which is the best programmable clock from Silicon Labs (.3 psec RMS jitter) [link]
- Clock drives the low jitter shift registers through a clock-fanout [link]. The jitter in the fan-out device is in the fsec range
R2R Module reclocker
- Clock is Si514 is the lower grade of programmable clocks from Silicon Labs (.8 psec RMS jitter) [link]
- Clock signal is transmitted through the FPGA for clock division and then to the shift registers. The added jitter in the FPGA is in the psec range
However, the reclocked signal in the R2R module feeds straight through the resistor ladder avoiding “several layers” of electronics as compared to a conventional implementation where the reclocked signal feeds a DAC chip.
In the end, the actual jitter as seen by the resistor ladder is the cumulative jitter consisting of following components
- Clock intrinsic jitter (0.8 psec RMS)
- Jitter added by the FPGA (I think in the order of 10s psec RMS based on datasheet numbers)
- Jitter added by the shift registers in the psec order based on general data on shift registers
10s of psec RMS jitter at the resistor ladder is pretty darn good in my opinion.
The onboard microprocessor is the STM32F030 uC [link]. It is responsible for:
- Measure input clock and program the Si514 programmable clock as needed
- Initially, volume control by using a potentiometer
- More features later since this is a general purpose uC
The specific device is the 32 pin device of the family with 16 general I/O pins. I believe some of the I/O pins are available through J1
- Designed to be powered by a single dual 7-8V, 5W transformer. Can also take an external +/- 7-15V DC supply. Filter capacitors are Nichicon 820uF 16V CL series
- “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….”
- A DC-DC converter (switch mode) provides the 1.2V for the FPGA core. Every other supply is low noise linear [link]
- The most critical supply is the +/- 4V reference for the resistor ladder. This is generated by a “two step, first to +- 5V, 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”
Here is a picture of the main supply section. The description is my best guess based on the information provided. I believe the digital section is powered by a DC-DC converter-regulator, except for the clock which has its own regulator.
CUSTOM FILTERS AND DIGITAL CROSSOVER
I think the ability for user-defined custom digital filters is a BIG feature for this DAC. In addition to the traditional DAC filters, one can load filters that implement crossover functions.
One of my frustrations with the ESS DAC is that I have not been able to take advantage of the custom filter facility. I am able to program everything else, except for the custom filters. Even though some claim that this feature works fine, I have not encountered any diy implementation and only one or two commercial implementations. Whether due to my own ignorance or to other factors (such as lacking documentation), fact is that there are no publicly disclosed diy successes of having implemented custom filters in the ESS DACs.
With crossover filters, there is finally a BIT PERFECT high quality DAC + digital crossover solution. More specifically, current digital crossovers if used with an external DAC of choice would add additional A/D or D/A conversions plus asynchronous sample rate conversion. Imagine a more “straight wire” implementation.
I have come to the realization that audio diy never ends. There is never the “final” project. There is always tweaking, a new mod, new stuff. Thus I decided to use a cardboard box as an enclosure. Much easier to work with, simple tools would do.
Many would consider a proper enclosure a must for a diy project, but for me this is good enough. Here is a nice looking Lenovo tablet box.
POWER SUPPLY TRANSFORMERS
Three separate transformers were used:
- DAC supply transformer: salvaged from an VCR from the time that linear supplies were still being used. These transformers are very nice, having at least 2 hefty 9V secondaries. The secondaries feed a TPA dual PS which is used as a pre-regulators for the DAC supplies. The two capacitors hanging off the transformer provides DC to an LED (to indicate power 0n/off)
- Analog supply transformer: 15×2 toroidal transformer. (Got it from Twisted Pear Audio)
- Controller transformer: 6V DC salvaged from a wall-type supply, upgraded the smoothing capacitors
Water bottle caps are perfect for the feet.
DAC AND ANALOG SUPPLIES
Used a quad-supply board. This board is designed to provide 4 x 3.3V/5V regulated output. Two supplies receives pre-regulated DC, two supplies receive AC (thus the larger filter capacitors).
Two of the regulators were modified to provide 14.2 V (1.4V+6.4V+6.4V) for the opamp by shorting pins 4 and 5 to GND. They are also configured to provide +/- 14.2V by tying the +output of one of the regulators to GND (I followed what diyinhk did for their dual +/- supply [link])
Standard build. Only “upgrade” are the electrolytic capacitors. On the backside, the jumpers are to connect the I2C lines to the separate I2C header because I forgot to short the lines on the front side before I soldered the connector (which blocked the jumpers). Later I will upgrade the opamp.
There is a separate SPDIF input connection that feeds the GPIO2 pin in the DAC. Selecting this GPIO pin for SPDIF input has been enabled in the code.
GPIO1 is configured in the board to be in input selection pin for manual selection of I2S/SPDIF. My code does not enable this mode because the selection can be done directly through the user interface. This is here for manual selection with a switch and requires that the chip be programmed in such a way. I believe the diyinhk XMOS interface would program the chip to allow manual selection of SPDIF input.
The board comes with an NDK 80 MHz oscillator [link]. Other implementations may use a 100MHz clock. The software support both 80 MHz and 100 MHz clocks.
In addition, a separate supply can be used to power the clock by cutting the power trace and connecting the supply to the through-hole vias.
ANALOG AND DIGITAL POWER
The external 5V powers a single regulator for the analog 3.3V AVCCR and AVCCL. . The second regulator provides 3.3V to in chip internal oscillator (I think in order to support an external quartz crystal instead of an oscillator). The regulators are marked “LLVB” and are TI LP5907 Ultra Low Noise regulators for analog applications [link]. They rank near the top among ultra low noise regulators [link].
The external 3.3V is used directly without further regulation to power the digital side of the chip and also the local oscillator.
Power to the oscillator is further filtered by a ferrite bead
Positive and negative supply voltage are taken directly to power the opmap.
The board has connection for the differential outputs straight out of the DAC chip and also single ended output through the opamp. I am using the single ended output wired to a mini-plug for connection to a headphone amplifier. The opamp provided is a NE5532 dual operational amplifier [link]
Nice, solid ground-plane
A new version of the code has been posted in the CODE tab [link].
This version has been fully tested with an Amanero USB interface [link] connected to the DIYINHK DAC board with an 80 MHz clock. Both PCM and DSD files of various sample rates were used together with foobar [link]
READ THE CODE CUSTOMIZATION SECTION
Make the proper adjustment for your specific implementation in the code.
/******************* Code Customization Section *********************/ /* First: Choose the clock frequency you have and comment the other */ #define USE80MHZ //#define USE100MHZ /* Second: Choose stereo or mono | CONFIGURATION | #define DUALMONO | #define STEREO | |---------------------|------------------|------------------| | Dual mono | un-comment | comment | | Stereo | comment | un-comment | |---------------------|------------------|------------------| */ #define STEREO //#define DUALMONO /* Third, optionally choose the number of inputs. 6 is the max without modifying the code. You could lower the number of input choices here. for example if you only want to see 2 choices, modify the code like this: #define ICHO 2 */ #define ICHO 6 /* Fourth, optionally change the name of the inputs. Keep 6 characters Use blanks if necessary. If you use less number of inputs, only the top ones matter. */ char no0 = "INPT-A"; char no1 = "INPT-B"; char no2 = "INPT-C"; char no3 = "INPT-D"; char no4 = "INPT-E"; char no5 = "INPT-F"; /* These inputs choices can be virtual or real. In the ES9018 there were 8 data lines. One could simultanously connect one I2S/DSD input plus 3 additional SPDIF input (thus 4 physical inputs). In the ES9018K2M there are two additional input lines for SPDIF so one can potentially connect one I2S/DSD input plus 2 additional SPDIF inputs.In addition one could choose different parameters -such as the DPLL bandwidh or filter selection- */ /* Fifth, adjust the interrupt routine to match your rotary encoder by adjusting the mode parameter in the following routine (search for it in the code): "attachInterrupt(0, rotEncoder, LOW);" The mode parameter defines when the interrupt should be triggered: LOW to trigger the interrupt whenever the pin is low, CHANGE to trigger the interrupt whenever the pin changes value RISING to trigger when the pin goes from low to high, FALLING for when the pin goes from high to low. You can also read the following link: http://hifiduino.wordpress.com/2011/09/12/problems-with-rotary-encoders/ */ /***************** End Code Customization Section *******************/
Here is the initial public release of the s/w control code for the NEW ES9018K2M DAC [link]. This project was completed at this (early) time due to the insistence of “syllable” at diyaudio who is running a group buy for his DAC board [link].
Download the code here [link]
The code is based on the ES9018 code and supports revision V of the chip (“E” marking in the third row of the text). It is also based on the 5/15/14 version of the official data sheet (available through authorized distributors and under NDA)
(As compared to what is available in the older ES9018 DAC)
- Support for a minimum phase FIR filter
- Support for separate DPLL settings for I2S and DSD (16 settings for each)
- Support for FIR filter (oversampling) bypass AND IIR filter bypass
- Exposed De-emphasis filters
- Enabled balance control in 0.5 db increment
I will be using the diyinhk implementation of the ES9018K2M (it is already populated and ready to use. I realize that I am getting to lazy to start with bare boards and some the surface mount chips I cannot properly solder).
The power supply is an older quad-supply board also from diyinhk which I’ve hacked to also provide +/14.x v. in order to power the opamp
Using my latest favorite Arduino clone (only $10) [link]. The two small boards hanging on the pins are a LCD backlight control and a 5V to 3.3V level converter. As I realize that this audio diy hobby is never ending, I figure doing things the easy way trumps doing things the neat way (like for example using an Arduino shield) so I soldered the wires directly on pins that I plug into the board.
Got an engineering sample of this new embedded board solution from iTead Studio. It is based on the Allwinner Technology A20 Dual Core SoC (The same processor as the Cubietruck board). iBox is being crowd-funded at indiegogo [link]. At $70 including power supply and case is an incredible deal.
The iBox is an example implementation of the modular approach that iTead is developing. A “system” can be configured with a “core board” and a “baseboard”. Thus iBox is a core board plus a baseboard and plus a case.
The case is made of gray-anodized aluminum with a plastic top and a steel bottom. (I added some rubber feet)
Here is compared to the size of a uSD card
Front side: uSD card reader, status LED and IR receiver
Side: Multi-function expansion connector
Back side: Peripheral connectors
The Core board
The A20 core board [link] is designed as a “computer on a module” and consists of
The core board is designed as a bare minimum computing module that breaks out most of the I/O pins and buses through two rows of pin headers. The approach also is to “standardize” the pin-header form-factor to allow mixing and matching with baseboards in order to suit different requirements. In addition, this approach provides an upgrade path to newer or different processors.
A20 SoC and DDR3 RAM (The GT chips, each 512MB). The 4 GB Flash should be in the back side of the board.
The Power Management Unit, AXP209
Detail Connection to baseboard
The baseboard in iBox is designed to provide peripheral interfaces and connect to a core board. The iBox baseboard is one of different baseboards that iTead is developing and as one of the first implementations, it aims at wide appeal by providing the most common I/O interfaces.
USB Hub: GL850G Hub
Ethernet Interface: Realtek RTL8201CP
The board has a 3 Amp switching regulator, the MP2307 set at 5V. The input range of the regulators is 4.75V to 23V.
uSD Card reader, IR receiver and LED indicator
Multi function expansion connector
USB and HDMI connectors
SPDIF Toslink optical connector, Ethernet and power connector. The bundle supply is rated a 9V, 2 Amp
Summary of iBox baseboard interfaces and connectors:
- Power connector
- 5V regulator (MP2307)
- Four 2.0 USB ports (Integrated GL850G Hub)
- HDMI port,
- Ethernet interface (Integrated Realtek RTL8201CP 100M transceiver)
- SPDIF optical (Digital Audio Output)
- U-boot button (Universal Bootloader. U-boot to embedded boards is like BIOS to PC motherboards)
- uSD care reader
- IR receiver (for remote control)
- Status LED indicators
- 32-pin multi-function expansion interface providing the most common interfaces
- Video output
- Serial Interface
- Debug interface
- SATA Interface
- Analog audio In
- Headphone Out
In fact there will be an expansion board [link] available with SATA connectors plus other connectors
Since this site is dedicated to audio, we will focus a bit on the audio capabilities of iBox
According to the datasheet, The A20 has the following built-in audio features:
- 16bit, 24-bit data
- 44.1KHz, 48KHz, 96KHz and 192KHz sample rate
- 100 db SNR
- Analog/Digital volume control (62 steps)
- Stereo headphone amplifier (capless). dedicated headphone output
- ADC: 24-bit, 8KHz to 48KHz, 96 db SNR
- Line-in Stereo or one differential
- Two Microphone input
- Stereo FM input
Here is the analog/headphone output diagram:
The iBox has a built-in SPDIF/Toslink connector. According to these discussions [link], the SPDIF output supports:
- 16bit data
- Up to 192KHz sample rate
- Resolution: 16bit, 20bit and 24bit
- Sample rates: 8KHz to 192 KHz
- Format: I2S, Left Justified, Right Justified
- Frame (BLCK): 16bit, 20bit, 24bit and 32bit
I’ve previously described the I2S capabilities of the A20 processor here [link]. The A20 datasheet (p.20) [link] specifies that the chip supports up to 8 channels of I2S output (DO0 to DO3 represent the 4 stereo channels of I2S data).
I2S support in the core board
In the Itead A20 core board pin schematic [link] we can see that the I2S pins are available and connected to the pin headers (PB5 to PB11):
I2S support in the baseboard
Looking at the schematic of the iBox base board, pins PB5-PB11 are not connected to the expansion header. However, PB5-PB11 pins are available on the underside of the base board (they are just soldered without connecting to anything) and can be easily tapped.
I shall get familiar with the software environment and report shortly in the next post…
ALLWINNER TECHNOLOGY, THE COMPANY
The A20 SoC was announced about a year ago. I have to admit, I had never heard of this company. A bit of digging uncovered that this company is fast becoming a dominant player in the SoC market:
You may never have heard of Allwinner but they are huge and as of CES now have an 8-core tablet part on the market. With the release of the A80 SoC and the OptimusBoard that SemiAccurate used, the company is well positioned for the mainstream tablet market in 2014.
Allwinner rarely makes the headlines because they don’t make bleeding edge products that go in to high-end phones and tablets, instead they make mainstream SoCs that go in to high volume tablets. This mid-range market has decent margins, huge volumes, and since they don’t target phones directly there are no radio hassles and regulation to deal with. How big is Allwinner? Huge. Continue reading: [link]
The most interesting part of this company is their announcement to join Linaro’s newly formed Digital Home market segment group as a founding member together with media behemoth Comcast (and others). This means that there will be more video and audio applications coming our way.
Linaro Ltd, the not-for-profit engineering organization developing open source software for the ARM® architecture, today at Linaro Connect Asia 2014 (LCA14) in Macau announced* that …
Allwinner Technology is a founding member of a new market segment group being formed in Linaro to focus on the Digital Home market. This group will be the third Linaro segment group, following the formation of the Linaro Enterprise Group (LEG), focused on ARM servers, and the Linaro Networking Group (LNG) focused on the networking equipment market space.
The list from iTead should work as is. Others are compatible with the A20 SoC, but may require additional work to support the peripheral components.
- iTead Documentation and Download Repository [link]
- Audiophile bit-perfect with the A10 [link]
- CNX Software Blog. Developments on embedded computing, including news on Linaro [link]
- Review of A20 built-in DAC and headphone output [link]