UADC4. Universal Analog to Digital Converter for Zero IF Software Defined Radios. (pre- launch during the AMSATSA Symposium 2018)

Transcription

1 UADC4 Universal Analog to Digital Converter for Zero IF Software Defined Radios (pre- launch during the AMSATSA Symposium 2018) INTRODUCTION Zero IF receivers or Direct Conversion Receivers (like softrocks, IQ mixers, WSE converters and IQ+ receivers) are the most popular SDR implementations used by Ham Radio operators after the Super Heterodyne Receivers. The introduction of PC audio sound cards creates an extensive platform for experimentation of which Ham Radio operators have profited in an incredible way for more than 25 years. But as usual, performance is a strong tradeoff negotiation; where the new PC audio cards cannot add more performance in our SDR RX systems, the answer is very simple: PC audio cards are just PC Audio cards ; they are not ADC systems designed for Software Defined Radios. Even if software developers are proclaiming software is everything, that is a wrong assumption because there exists a lot of space for HW improvements. Software cannot fix what HW cannot deliver!!!

2 The adoption of PC audio Cards as ADC systems is just one big example of the invention and creativity of Ham Radio operators within the last 20 to 30 years, but it is not the final answer for a correct Analog to Digital Conversion in a Ham radio SDR system for Weak Signal Communication like EME. It s just a cheap solution running on its limits since many years! The UADC4 is not a PC audio card, it s far away from that category and cannot compete with professional audio cards; in the same way those cards cannot add any single bit of performance in your Zero IF SDR. The UADC4 is specially designed for SDR applications and if tested as an audio card its results will be not as good as the commercial audio cards, but in conjunction with your SDR receiver it will create a robust platform with outstanding performance where the limiting factor will not be any more your PC audio card used like and ADC; that limiting factor will move quickly to your Local Oscillator or your mixer, areas where many options exist to create a high performance SDR receiver with much more performance than traditional SDR receivers based on simple PC Soundcards. This document is an explanation of what I did, looking for a better performance for my IQ+ radio, creating a dedicated ADC with some special characteristics you will not find in any commercial PC soundcard. Are ZERO IF RECEIVERS old fashion technology? Many people think YES, they are! But with all respect my respect, I think they are wrong!! Hardly Wrong!! The market is inundated by DDC receivers (Digital Down Conversion Receivers). The most promising argument for their use is: DDC SAMPLE ON THE ANTENNA. THE REST OF THE JOB IS A MATTER OF SOFTWARE. That statement looks convincing, but looked at from the Ham Radio perspective is far away to be 100% reality, because there money counts and therefore plays a decisive role. For sure DDC Receivers have extraordinary advantages in fields where Zero IF Receivers cannot compete. For example, the use of pure software NCO (Numerical Computer Oscillators) fully implemented in a FPGA by software, eliminates the tedious problem of phase noise and reduces the reciprocal mixing problems dramatically, but how many of those radios really come up to this well-known promise? Only few ones and those are too expensive for a Ham budget. Some problems with DDC receivers are related to sample capacity, the dynamic range they can really deliver and the excess of bandwidth they can sample. Most of the commercial application chips today can sample no more than 16 bits and few ones can go up to 18 bits in frequencies below 50 MHz; chips capable of doing real DDC conversion over 100 MHz are rare and complete out of reach for a Ham budget; they are mainly designed for military purposes. Those with high frequency response have less than 12 bit resolution.

3 Just reviewing theory: In a digital system the maximum theoretic Dynamic Range is defined by the following formula: DR = (6.02 x N) - 3 Where N is the number of bits your ADC can sample, in this way: A 14 bit = (6.02 x 14) -3 = 81.28dB A 16 bit = (6.02 x 16) -3 = 93.32dB A 18 bit = (6.02 x 18) -3 = dB A 20 bit = (6.02 x 20) -3 = dB A 24 bit = (6.02 x 24) -3 = dB But those figures are just pure theory and in real practice, far away of reality; a rule of thumb for a 24 bit system is: Decrease theoretical value by 20 db, and then you will get a respectable 121 db Dynamic Range. This is only an example with one strong signal in your pass band; as soon as you have more than one, the SUM of those signals cannot exceed the saturation point of your ADC, thus decreasing the effective Dynamic Range of your system. But this is not the case with most DDC receivers, where the ADC of those receivers is hardly sampling at 16 bits. Now compare and see how much Dynamic Range you lose just for sampling on the antenna a big amount of BW? And as soon you are forced to use Analog Down Converters with mixers and Local oscillators to place the high frequency on the spectrum BW of your DDC, the promise of sampling on the antenna just vanishes, introducing all the behaviors of traditional receivers with mixers. We are in a transitional period where hybrid systems (DDC + Zero IF Receivers) are gaining popularity due to the absence of affordable DDC chips over 100 MHz. Zero IF Receivers may be old, but are still alive, and they will remain useful for many years. Pure 24 bits sampling on frequencies above 100 MHz is only for DDC chips for military and commercial applications, because the cost per chip may exceed 5,000 USD, and they are not available in low quantities. But even with the poor dynamic range they can deliver and they are able to sample many Megahertz in a fraction of seconds. Is that really what you want for Weak Signal Communication like EME, Satellite or shortwave receiver? An economic and affordable solution is to use Direct Conversion Receivers with a properly designed ADC and NOT with a PC sound card acting as an ADC. DDC receivers and Zero IF receivers were subject of many comparisons within the last years: Leif SM5BSZ did a professional comparison, testing 3 DDC well-known receivers (Perseus, SDR-14 and SDR-IP) showing a good dynamic range. A similar comparison was done by Loftur TF3LJ/VE2LJX on a Softrock 6.3/Mobo/SDR-Widget Lite with a 24 bit audio card. Both comparisons showed very similar performances

4 and both systems demonstrated better performance than a traditional Super Heterodyne Receiver. ABOUT DYNAMIC RANGE Many times when I spoke about Dynamic Range within the last EME conferences some people told me. Dynamic Range is not a problem for me. and my answer always was good for you being a privileged EME operator. Dynamic Range is not the only value you analyze in a RX system but it s one of the most important, especially in a congested world, were more and more Ham Radio operators locate antennas in cities and densely populated areas with tremendous man-made noise sources near/around the antenna. Just few ones have the privilege to run EME antennas located in remote areas where a high Dynamic Range will not add any value to the system due to the almost absence of strong signals. Just to have an idea, here in South Africa, in the Karoo Valley exists the SKA (Square Kilometer Array) radio telescope, equipped with hundreds of 12m offset dishes (when finished), running from 1 Ghz to 5 Ghz. The government, to protect this world wide project, created an exclusion zone of several square kilometers around the antennas site where all single town, farm, industry, power line cable; train railways were removed and relocated, away from the exclusion zone. Even airplanes when they approach the site are forced to down transponders and many electronic devices like weather radars, etc. You cannot visit the site carrying any electronic device like 2way radios, cameras, laptops or mobile phones they are strictly forbidden why?? Well the DDC converters on those antennas sample at very high speed and at high frequency, producing terabytes of data per day, BUT with an ADC sampling at just 8bits. Means very poor Dynamic Range!! But they don t need more due to the fact they run the installation in a protected area. - Is your antenna located in a similar place? Dynamic Range is very important and becomes more important every day!! LIMITING FACTOR ON PC SOUND CARDS FOR ZERO IF RECEIVERS As I mentioned initially, PC sound cards are just PC sound Cards and not Analog to Digital Converters for Zero IF SDR radios; they are designed for home or professional audio applications like voice and music in a very limited band width. PC Audio Cards, even if they can sample up to 192 KHz, are confined to their exceptional specifications in just few Kilohertz, from 20 Hz to 20 KHz to be precise. The reason for that being human brain and ear response. Nothing above 22 KHz can be heard by human beings. So it doesn`t make sense to extend high tech

5 specifications to a spectrum not possible to be heard by the consumers. Nevertheless the audio industry is pushing sampling rates up to 192 KHz with the aim to reduce noise. This is one of the most polarizing topics within audiophiles. In the late 80 s clocking was one of the biggest problems for the primitive PC audio cards; with high jitter those cards introduced many errors in the ADC process. That was the time of 44.1 KHz sampling; those cards also suffered from poor anti-alias filters used to remove inaudible supersonic frequencies to avoid artifacts mask, harmed or even destroyed the audible spectrum. Then the move to 48 KHz didn t prove to be a solution until better filters were available, and the supersonic intermodulation distortion was not a problem anymore: A little increment on the sampling from 44.1 to 48 KHz, associated with better clock and filters resulted in better audio cards. The extensive use of Super High Frequency Equalizers, Superfast Compressors and Electronic Synthesizers (That was the boom time of the Electronic Music) justify the introduction of oversampling moving fast to 88.2 KHz, when most PC Audio Cards started using oversampling for DSP applications later were extended to 96 KHz BUT always with the limiting factor of the human brain and ear, just with 20KHz BW at the end. Today 192 KHz sampling in PC audio cards is getting more popular but still under development and those cards sampling with limitations at that frequencies. The reason for that: again the human factor. Why the industry will develop chips with better flat response above 20 KHz if the limiting factor is the human brain and ear? Waste of money and for sure SDR applications were never taken into consideration. Today many small Audiophile houses are producing converters (AKA audio cards) with just 44.1 to 48 KHz sampling with outstanding performance, killing the most reputed PC audio cards brands sampling at 192 KHz. I leave that discussion to the audio professionals, but it looks like having a human condition as a limiting factor, expressed in just 20 khz of BW, making it useless to produce high quality converters for higher sampling rates. If you have access to a 192 KHz sampling Audio Card you will realize that the frequency response is terrible; most of those audio cards keep a flat spectrum in just 20 KHz (the human limit) and after that the noise rolls up to 20 db in the corners. Looks like here we were confronted with the ordinary idea of more is better, like such analogies of more pixels would result in better videos, or more sampling speed in better audio or more Megahertz in your clock computer would create faster computers. Intel, after sales number decreased, quit the widely spread idea of more Megahertz leading to faster CPU s, instead of that, Intel moved to a moderate clock speed increasing the numbers of cores per CPU, which creates really fast computers. But at that time the CPU industry was governed by the idea of More Megahertz more sales. In the same manner today exists a misconception about DDC receivers: Sampling on the antenna big amount of BW, because more band width is better. Completely wrong from the perspective of EME and Weak Signal Communications.

6 LIMITING FACTORS OF EME SYSTEMS We can create a list of all subsystems constituting a typical EME station. The interaction of all of them will rule the final performance of the system and any improvement of one single subsystem will not necessary produce better overall performance. The limiting factor will move and jump from one side to another as soon as we start improving the performance of some subsystems: performance is like a moving target. Here a simplified list of common subsystems of an EME station, starting from the antenna to the working position: - Antenna (including a tracking system) - Pre-amplifier - RX cables and relays - RF Power amplifier (and TX coax lines) - Receiver (Transverter, superheterodyne, SDR, PC Sound Card) - Computer (with the associated software) - Operator experience Now think: If with that system we are unable to hear our own echoes, an easy approach will be to increase the RF power 3 to 6 db, or we double the size of the antenna, or we optimize the feed for better gain and/or lower noise, or we replace a lousy preamp with 0.7dB NF for a good one with just 0.2dB NF, or just replace a single piece of RG-8 in the RX system before changing the preamp for a better one. Or we change all of them. Many areas, different approaches, but now think if you just replace the radio; instead of and old RX receiver you decide to install a new Icom IC7300 with all the latest technology in his excellent receiver, BUT you keep all RX subsystems intact. Your antenna system has two sets of antennas, one for Vertical and one for Horizontal and you switch the RX from V to H manually. The result will be in no way better than with the previous system, and you already spent good money for the new radio for NIL performance increment. Instead of that, for just less than 100 USD, you can install two Softrock receivers and setup Linrad in your PC, then you will defeat faraday rotation running a properly Adaptive polarization system. The performance increment will be tremendous for just 100 USD with little effort compared with the expensive IC7300 who will add nothing to your system. Optimization doesn t need to be expensive in terms of money; all you need is to be clever enough to identify the weakest points on your system and place the money on that areas. Just a drop off replacement is not always a good tactic, especially in systems with already medium to good performance. If you already have an optimized system and you want to go for every single subsystem to its maximum, then a titanic job is in front of you, especially when you come to the Receiver subsystem.

7 Now think: If your system is already well optimized, you run a good antenna array, let say 4 long yagis for 144 MHz, good preamp, cables and so on, and you are using a RTL or FDC dongle to feed MAP65 and constantly note a big amount of interference in your pass band, sometimes in specific directions, probably you are running short in Dynamic Range (the USB dongles have moderate Dynamic Range and they overload very easy). After a proper assessment you definitively identify your problem stems from a lack of Dynamic Range, you will need a better SDR with 20 or 30dB more Dynamic Range to solve the problem. WHY PC SOUND CARDS ARE NOT SUITABLE FOR ZERO IF RECEIVERS FOR ADC CONVERSION? Well, I will said they are but with limited performance. In the initial part of this document I explain many reasons for that, inspired by many documents and public information, but for better understanding I will invite you to read what Dan Larvy wrote about high sampling and audio cards. He is one of the most respected designers of audio converters in the world and a die-hard opponent of ultra-high sample rates. His observations are limited to the audio spectrum, but we can easily understand that the same theory governs the world of ultra-high speed DDC Receivers. The Nyquist Theorem is valid independent of the frequency and cannot be ignored. (here the link) In resume Dan Larvy wrote:. Nyquist pointed out that the sampling rate needs only to exceed twice the signal bandwidth. What is the audio bandwidth? Research shows that musical instruments may produce energy above 20 KHz, but there is little sound energy at above 40 KHz. Most microphones do not pick up sound at much over 20 KHz. Human hearing rarely exceeds 20 KHz, and certainly does not reach 40 KHz. The above suggests that [even] 88.2 or 96 KHz would be overkill. In fact all the objections regarding audio sampling at 44.1 KHz are long gone by increasing sampling to about 60 KHz. Now, how we can pretend PC sound cards are suitable for SDR receivers when they are just well designed for 20 KHz? The excess of sampling capabilities in PC Sound cards arrive with the first 20 KHz of response well designed but with the rest of the BW just like a noisy add-on with several limitations. We used and still using PC sound cards to do the ADC conversion because they were the only affordable solution for decades, those expensive Audio cards still dominating nowadays, and they are a serious limiting factor in Ham radio SDR systems.

8 For that reason, I decided to look for a proper ADC to do the job and keep audio cards to provide just PC sound, Skype or hear music. I already retired my PC sound cards from the Radio station. UADC4 DESIGN I started the process 2 years ago. Having no experience at all in ADC design, in the beginning I was asking myself where to start? Definitively any attempt to design my own ADC had to follow the basic rules of AD conversion, and learning those rules was a tedious job with extensive reading and testing, where Leif SM5BSZ came in and played a key role teaching me many aspects about AD conversion. After understanding the principles of the AD conversion, I was looking for the proper AD chip and was confronted with the reality about how difficult the selection could be. Another important point was how to stream the digital data into the computer without invest too many efforts in extensive programming. The use of FPGA chips was discharged due the complexity and cost. Due to the low frequency involved, I finally was forced to look for Audio ADC chips with all pros and cons already explained previously. I finished with the following 4 types/labels: - Analog Devices - Texas Instruments - Linear Technologies - AsahiKASEI I choose AsahiKASEI who has proved experience in ADC chips with outstanding (audio) performance. The selected chip, the core of the UADC4, is the AK5574, a 4 channels Differential 32bits ADC up to 768 KHz sampling capable of deliver up to 121 db Dynamic Range in single mode and up to 127 db in 4 to 1 configuration.

9 I decided to use the AK5574 based on his flat response above 20 KHz (where most of the audio chips have a terrible rollup in the noise floor). Compared with other ADC s, the AK5574 has a good flat response above 40 KHz up to 96 KHz, after that increase, the noise floor rolls up fast to 192 KHz. Above 192 KHz the chip suffers the same flaws as other brands, which means that the noise floor is no more flat at all. For sure the use of a preamp in front of your SDR receiver and the proper Linrad calibration will take care of any noise floor unflattening, but I was looking for a device with better characteristics in its native configuration. The AK5574 can select the sampling speed without any special programming, the chip locks its sampling speed according to the software request you are using to open the data stream, in my example Linrad opens different sampling speeds up to 192 KHz with no problems, also MAP65 opens data stream at 96 KHz with no programming intervention, HDSDR in the same way. The output data format supports in PCM mode 24/32-bit MSB, justified, I²S or TDM; in DSD mode it supports DSD native 64, 128, 256. Additionally, the AK5574 has internal anti-alias filters; compared with other chips those filters have a good response. You can configure the PCM roll-off filter response with one jumper between sharp and slow roll-off and you can configure the PCM digital filter delay between short and normal. In the best configuration at fs=96 KHz the passband is selected at 48.8 KHz with -82dB Stopband KHz.

10 THE BUFFER STAGE by THE MANUFACTURER After the ADC chip was selected I tried to understand why almost all ADC chips available for that frequency suffer from asymmetrical noise floor. I did measurements in many audio cards with different ADC s and I discovered peculiar buffer stages in the input. Independent of the internal configuration of those chips, where apparently they concentrate the perfect response in just 20 KHz (other ones, like the AK5574, extend excellent performance up to 96KHz) the buffer stage has a lot to do with the final performance. Looking into the manufacturer s documentation, this is the buffer stage proposed for the AK5574 by the factory: This is definitive a buffer stage not suitable for our purposes, it is okay for audio and music but has many pitfalls for a SDR application. - The NJM5534 is an opamp with and input noise voltage of 3.3nV/ Hz; this can be improved with a better opamp. - The AK5574 needs differential input but this buffer is not really a differential amplifier, I did hundreds of hours of simulations in LTSpice and the output amplitude, in this configuration, it has an asymmetric amplitude in the range of 80 mv when measured in pins AIN+ and AIN-; to properly feed the ADC, both signals need to have the same amplitude, no differences at all are accepted in this respect. - The first opamp causes a delay in the signal, as you see, when JP1 and JP2 are in place, the output of the first opamp goes to the input of the AIN+ opamp, the third opamp receives the signal without any active device, direct from the input source, even if the first opamp has his source resistor and feedback resistor in a ratio of 1 there exist a delay in the signal, something unacceptable for a SDR application. - The values proposed for the source resistors and feedback resistors are too high. The higher the values, the higher the noise caused by the resistor (thermal noise), but it looks like the noise produced there still within the parameters of what is acceptable for audio applications, but not for SDR.

11 - Last at not least, this pseudo single end input to differential buffer has no limiting control in the input, probably it is open to the designers to add some kind of potentiometer or digital resistor in the input to control the amplitude from the frontal panel or via software: see the power supply for the opamps, is +-15VDC, means the output of the opamps can swing the entire rail to rail voltage, thus the risk to overload and kill the ADC is very high. For sure you can add, as I said, some kind of amplitude control in the input but this will add more noise to this already noisy buffer. THE UADC4 BUFFER STAGE (Sorry if I not disclosure the diagram because we are having hard times with clones and illegal copies on ebay and mainly from Asia, the last scandal is what happens with the mchf SDR radio, have a look in Google.) To correct all this problems in the buffer stage I decided to build a complete new buffer suitable for our SDR demands: - I discarded the NJM5534 opamp and selected two new devices from Linear Technologies with an input noise voltage of just 1.0nV/ Hz - The first device works as a limiter to avoid any possible ADC destruction due to an excess of amplitude level. - For the second opamp I selected a real differential amplifier to correct the asymmetric output and incorporated a LPF in the design to reduce frequencies above 100 KHz - Feedback resistors and source resistors are selected properly to reduce the noise caused by those resistors. - The resistors are high quality parts with tolerances expressed in ppm, the tolerances here are better than 0.2% - Avoid the use of any kind of potentiometer or digital resistor to control the input amplitude; instead of these noisy elements (very common in PC Audio Cards) I deployed a very simple set of jumper matrix to accommodate input values from 0 to 14 Vpp in 5 steps. - To set up the proper input gain, the user will just hard wire the input using jumpers and close the unit, for example with the jumper in position 1, the maximum input will be 3.5 Vpp producing 2.8 Vpp in the output buffer. If more level is applied in that configuration, the first opamp will saturate, the senoidal will be distorted but the amplitude introduced to the ADC will never exceed the 2.81 Vpp, preserving the ADC from any damage. THE CLOCK The UADC incorporate a very precise and low jitter clock, one common problem with PC Soundcards is the clock quality, normally the clock units are very cheap oscillators with very high phase noise and poor stability, no so critical for music but very critical

12 to keep the reciprocal mixing as low as possible in your SDR. Everything with a phase noise high than is not good enough to keep reciprocal mixing as low as possible. The internal clock in the UADC4 is the GXO-7506, it s an Ultra-low jitter Oscillator, some professional Audio cards has this unit but not so common in consumer PC audio cards. At 1KHz the GXO-7506 has and extraordinary performance with just -155dBc/Hz, at 100KHz separation the phase noise is just -174dBc/Hz. The phase jitter RMS (12KHz to 5.0 MHz) is in the range of 0.04ps (pico seconds). The ADC As I mention previously the fact we are dealing with base band signals in the range of 1Hz 192KHz force me to use a commercial ADC used in audio applications. The selection was rigorous and different candidates was tested, as I mention previously the AK5574 from AsahiKASEI accomplish all my expectations, specially respect to the flat responds beyond the traditional 20KHz umbral, up to 96KHz the noise floor response is very flat with and small increase up to 192KHz. In the beginning of this document I mention the most relevant information. If you want to read detail information about the AK5574 here the link: Internal power supplies The UADC needs different voltages to work properly, the unit is feed with 11 to 14.5VDC but internally different voltages are feed to the different stages. Linear regulators are not in consideration due the high noise they produce at different frequencies. The voltage regulators are Switching style and based in very well tested circuits in ADC conversion by Linear Technologies, the use of switching regulation is not easy but when is well implemented they deliver good level of current with almost no noise, ADC systems are sensitive to ripple and high oscillation noise coming normally via the DC path. That condition demands the use of high quality capacitors with extremely low ESR and properly inductors on the switching oscillators, additionally every single DC path is hard decoupled to the respective ground plane to avoid artifacts via the DC path. The DC decoupling demand a rigorous separation between the different grounds planes to avoid mixing the digital ground with the Analog ground. This separation is translated in a mandatory separate ground on the pcb and the result is a pcb design with 4 layers and not 2 as most of the consumer audio cards are using.

13 The PCB layout If the success of an electronic design will be just pick up and place the right parts would be very easy to produce high quality devices, no other part has more impact in the performance like the PCB layout. First, we need to follow the recommendations, specially by the ADC manufacture and second, we must avoid interferences from the different signals going back and forward within the PCB. The UADC PCB is a 4-layer PCB with microscopic vias passing through the different layers, most sensitive part is the definition of the Analog Ground and the Digital Ground, they must be isolated BUT at the end connected in just one single point to avoid ground loops. A 4-layer PCB is not easy to build, demand a lot of testing, in that way the PCB layout was tested 4 times before we arrive to a properly design. One of the most difficult problems to resolve is the magnetic interaction between traces, a poor isolation will produce signals become induced to other circuits where they don t need to be, reducing the channel separation and increasing dramatically the reciprocal mixing in the SDR. This is one of the most common problems with PC audio cards and why they don t reach high level of performance. This magnetic induction is very sensitive and complicate to reduce, specially in the input buffer, just moving a capacitor few millimeters in one direction could represent better or worst performance by several db s. some basics rules in PCB design are follow but at the end is a tedious try and error job with the obviously impact in cost and time. This is proto1 PCB, was discharged due magnetic loops in the input stage

14 The XMOS USB interface One thing is convert base band signals to digital and another thing is to place that data available in a format easy to manage by computers and software. We evaluate different kind of interfaces, like ethernet, SPIDIF (optical) and USB. We found the USB interface a convenient way to do that job, Ethernet was not considered due possible timing problems (especially affecting Adaptive polarization receivers) and for sure the complexity, optical interface was not considered for cost related issues and popularity, never the less we are talking here a moderate band width. The USB port present an excellent interface and for that to speed-up our design phase we decide to use the DXIO32ch produced by DIYINHK and based on the XMOS chip XU , with 2000MIPS in dual issue mode can work up to 384KHz with a incredible performance and ZERO impact on the signal quality, just demand USB.2.0 and the driver deliver with the board allow up to 4 analog channels up to 384KHz, 6 channels up to 384KHz in I2C and up to 8 analog channels with the properly firmware, an over kill board but at extraordinary convenient price giving top of performance. We just plug the XMOS board into our prototypes and the digital stream from the ADC was introduced in the PC with out invest time and money in a proprietary design. The board is deliver with the properly ASIO driver, an excellent way to manage low frequency data with extremely low latency.

15 UADC4 Design process, Performance and production During the initial phase of this project I start with the assumption than PC Audio cards are the limiting factor for better ZEO IF receivers, the numbers don t lie, a 15 to 20dB increase in performance was expected and the initial test demonstrate and confirm my hypothesis. The UADC4 born as a decentralized design, done in Canada, Sweden and Switzerland with central coordination in South Africa, the Canadian part was operated by George Boudreau, a well renowned Electronic designer with a lot of experience in SDR radios, he produce the famous Widget SDR radio and several DAC's like ODAC and USB interfaces for audio, in Sweden Leif SM5BSZ place all his experience for more than 50 years into the UADC4, Leif is a well reputed and rigorous Engineer who design the famous SDR WSE units many years ago and he is the worldwide standard lider for SDR HW performance test, he develop the best SDR software "LINRAD", last at not least I m responsible for conceptualization, the prototype production and performance test together with Leif, using all my experience accumulated within the last 10 years, we are very proud about the results with the UADC4. I want to emphasis Leif has no commercial links in any way with the UADC4 or LinkRF, he is just interested in high performance hardware, any endorsement from him is just because the UADC4 cover his expectations, more details you can obtain in an independent way direct from him, he can speak freely about how good or bad the UADC4 is, if you are interested contact him direct for his personal opinion and follow the test in his YouTube channel. At the end of the document you will find a dedicate table with all the data collected by Leif SM5BSZ when he compares many PC Soundcards with a Softrock SDR as a ZERO IF mixer in front. The UADC4 is located at 1 st place with outstanding results and the only reason numbers didn t go even better is because now the limiting factor is the ZERO IF mixer. A better mixer will produce much better results, like you can see the performance reached by the WSE units in one of Leif videos, the WSE units was designed 20 years ago and he never reach that level of performance until now when he use the UADC4 as dedicate ADC. Here some statements done by Leif after the last UADC4 proto was tested by him: Mainstream development is towards direct sampling in the VHF range and many regard units like the Softrock Ensemble more like a toy than a real radio. Major improvements are possible however. Alex, HB9DRI, is working on a 4 channel ADC for usage with direct conversion SDR. The noise floor should be something like 15 db lower than the best soundcard in the table below when used with a Softrock having the Si570 replaced by a good crystal oscillator. The WSE RX2500 was designed in year 2001 with components of that era. The mixer uses 74HC4052 with about 70 ohm switch resistance as compared to 4 ohms in the FST3253 used in the Softrock Ensemble. Yet, mixer performance is dramatically better as you can see here: uadc4wse Noise is 8 db lower and harmonics -70dB as compared to -45 in my most modified Softrock. What I see is reciprocal mixing noise floor@10khz = -156 dbc/hz. That is something like 10 db better than the

16 Perseus. It uses the UADC4 much better than the Softrock Ensemble, but improvements are still possible. The very sharp anti-alias filtering provides some noise and a too high gain. By running the UADC4 at 192 khz one could move the anti-aliasing into the digital domain and get a significantly better performance. A low Q anti-alias cutting at 96 khz would not have to add gain or noise. Further, the UADC4 provides some filtering above 96 khz. Popular trends today say direct sampling is SDR for the future. The UADC4 demonstrates that direct conversion today can be better. The direct conversion architecture of Softrock would need a lower noise LO, a better mixer and good front end filtering to become better than the best direct sampling radios. They also need good RF filtering.....end of quote As soon the last proto arrive in South Africa (June 2018) I will complete the test and publish definitive numbers, before that I want to be cautious due the fact we are testing now the automated production, if all test pass ok that last proto (Nr.4), will be the final unit for mass production. Never the less, the UADC4 move the limiting factor from the AD conversion to the mixer, new ZERO IF mixers need to be re-design to exploit the maximum performance and that ZERO mixers are already under design on my bench now!! Stay tune!! 73 de Alex, ZS6EME (HB9DRI) hb9dri (at) emeham (dot) com

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