New page is up, it’s my personal Linux Cheat Sheet, with somewhat of a bias towards audio streamers : https://www.dimdim.gr/linux-cheat-sheet/
New page is up, it’s my personal Linux Cheat Sheet, with somewhat of a bias towards audio streamers : https://www.dimdim.gr/linux-cheat-sheet/
Hi guys, long time no see.
Easter time here, so I have some time to spare, so here it goes.
Update on my DAM build
My Soekris has had a number of upgrades performed to it over the last 2-3 years that I haven’t posted about and it’s about time that I did.
The first upgrade was a switch from Salas BiB 1.1 to an UltraBiB 1.3. I wrote about the upgrade on the official UltraBiB thread:
I was (and still am actually) one of the lucky few that got to beta test Salas’ new baby.
This thing is remarkable.When we swapped it in place of the BiB 1.1 in my Soekris, the improvement was immediately obvious and not subtle. There was a general improvement in clarity and silence, but the biggest improvement (imho) is that the music appeared to have more energy in the lower mid area, where before it was kind of “dry”. This was with Salas’ very first prototype, built with standard (non-boutique) components. The board that I built with audio grade capacitors in the filter bank and MUSE BP caps in the output sounded even better.
I do need to experiment further with different brands of caps (especially in the C2 & C3 positions) but in any case this is an excellent power supply, substantially better than the BiB 1.1, both subjectively (the way it sounds..) as well as objectively (measured performance).
The next upgrade was when I upgraded my firmware to the newer firmware (rev.1.21) that doubles the number of supported filter taps from 2K to 4K. Even the basic / included filters provided a substantial improvement in imaging and detail. Definitely a worthwhile upgrade.
The final upgrade was the design and implementation of my own XMOS-based USB receiver, specifically for the DAM:
The idea was to include just the right mix of features that would make sense for the DAM1021. So, I did include ultra low noise LDOs (LT3045 & LT3042), SDA oscillators, reclocking with a Potato FF, but I skipped galvanic isolation since the DAM already supports it.
The improvement it made in sound quality was remarkable. It’s funny how all digital devices that are supposed to be immune to I2S signal quality turn out to be.. well.. influenced by I2S signal quality. The USB receiver that it replaced was the DIYINHK XMOS receiver that I have used in my DAM, which was pretty decent at the time of purchase. But the newer XMOS family (XUF208) coupled with the ultra low noise LDOs and the reclocking apparently did the trick.
The funny thing is that the prototype I built was actually tested in a good friend’s DAM and has been living there ever since. Someday I’ll build another one for my DAM. For the time being, I’m busy with my AKMs and ESS’.
Update on the DAM board offering.
A few days ago Soren announced Rev. 7 of the DAM1021:
According to Soren, the new boards will include the following changes / improvements:
1) it use the same FPGA and same uC as usual, so should be able to use same firmware, there are some differences, so new firmware are needed and will autodetect board.
2) The shift registers are now also in the smaller qfn packaging as used on other boards, so I don’t need to stock the tssop types anymore….
3) Output shift registers now running at 1.4M/1.5M sample rate, making FIR2 filter different. Will require new filter file, but plan to use same filter file for both old and new boards, firmware will load filters as needed.
4) The R-2R network is now 24+3+1, ie 24 bit classic R-2R network, 3 bit thermometer network, plus the sign, total 28 bits.
5) Yes, still same Si514 clock, but now in a small package (like I use on other boards), and it drives the shift registers directly, no added jitter by going though the FPGA. People who want to bitch about the Si514, please go elsewhere, we have heard you multiple times….
6) The clock Power supply is separate and use a LP2907 low noise regulator.
7) New vref supplies, like used on the dac2541. For those who want details, yes it use a opamp with transistor buffer, with a polymer output capacitor. Lower vref impedance from DC to Megahertz.
8) Those are powered at +-5.4V from LM317/LM337 pre regulators.
9) Output opamps are opa1678.
10) Power in are now DC only, optimal is +- 9V.
11) The board is mechanical and almost electrical compatible with previous version of the dam1021.
12) J9 is 4 bits user i/o, planning to use it for additional SPDIF inputs.
13) The non isolated serial management port is now TTL levels, not real RS232 levels. So get the right USB to serial adapter….
So quite a few changes, with the most significant being the Si514 having its own regulator (mentioned at a later post in the thread) and driving the shift registers directly. Plus new VREF supplies.. These improvements should be pretty audible.
It looks like the only flavor that will be available will be the dam1021-12.
We’ll see how it will go.
Archphile used to be the RPi audio distribution of choice for the most hard-core RPi-lovin’ audiophiles around until last June when Michael (tuxx) decided to drop it.
Then in January he hinted to it coming back at some point.
Well, it appears that that moment has come. Head over here for more: https://archphile.org/blog/archphile-119-codename-corona-for-rpi23-and-odroid-c2-is-up/
And not a moment too soon.. My Audio Pi is pretty much ready and it is in desperate need of a proper audio distribution. This should be interesting.
It’s been two and a half years since I posted about my Dual Mono AK4490 DAC. That DAC has been built and is in use by at least 6 people, other than myself. All of them have been very satisfied with its performance.
But since then quite some water has passed under the bridge.
Among other things, the AK4493 chip came out, and it was just that much different than the AK4490 that I had to update my design to accommodate it.
Feature-wise it was pretty enticing. It looked more like a limited performance version of the AK4497 than an upgraded version of the AK4490. So I had to try it out.
Since I was going to update the PCB design, I thought I might as well improve on as much as I could. So, the new board would:
In addition I would use the then new Si544 programmable oscillator, offering improved performance over the Si570. This did not require any changes to the pcb.
This is the updated schematic:
This is the 4-layer PCB:
And this is the BoM (v1.9) in xls format: Dual AK4493 DAC (main board BoM) (3931 downloads)
The finished board looked like this:
The design consideration, powering scheme and clocking considerations remain the same as with the original design. There is not much sense in repeating the same text here. I will make a few notes though, based on the experience gathered from building, testing and listening to several such DACs.
Also, having a properly designed and implemented USB to I2S receiver is very important. Early on I realized that it would be best if I designed my own XMOS-based receiver board, custom tailored to my needs. It would also include some light USB line conditioning and an AK4118-based S/PDIF receiver with 4+2 inputs. I would then standardize my DAC designs with this inputs board in mind, including properly supporting it in my Arduino code.
And so this board came to be:
Describing in detail this board is beyond the scope of this post, but suffice it to say, building it is not for beginners. Plus you will need XMOS’ xTAG programmer to burn firmware into the XMOS chip. If anybody is feeling particularly adventurous, drop me a line and I’ll see if I have any PCBs left.
Now, regarding the Arduino code needed to control this board(s), it is not very different than that used to control the AK4490s. The main differences are:
There are a number of to-do’s though, such as displaying the bit depth of the incoming PCM signal (from the USB port), plus more information on the incoming DSD stream (such as whether it is in DoP or Native format).
The hardware of the controller is the same that was used with the Dual Mono AK4490 DAC.
In this download I am including the modified versions of the libraries (as mentioned in the above linked post) as well as the necessary font files. Be sure to extract the contents of “Libraries (place in Libraries folder)” to your Arduino IDE’s “libraries” folder.
Download it here: aKduino v3 (7270 downloads)
Regarding the output stage, it is the same design that was used for the AK4490 DAC. However its output level is slightly lower than that of the AK4490 board since the AK4493’s VREF voltage is limited to 5.25V, compared to about 7V of the AK4490. This difference in volume is easy to compensate for by changing a few resistors on the output stage.
If anybody is interested in building this DAC drop me a line. I have a few spare boards lying around.
First off, I’m ashamed to admit that I had this little gem in my possession for about 2 years before I finally got the chance to put it through its paces.
After all, it’s just a s/pdif output device for a Raspberry Pi, right? I mean, it’s just s/pdif, how good could it be?
It turns out it can be pretty damned good! But I’m getting ahead of myself.. Let’s start at the beginning.
The DigiOne is a HAT compatible with most if not all RPis and supported by most if not all audio distributions. It is intended to be plugged-in directly on top of the RPi, with no need for an isolator HAT. Plus, it is designed to be powered by the RPi via the GPIO header, so no need (or provision) for an external power source.
The DigiOne utilizes a WM8805 to convert the RPi’s I2S signal into s/pdif. The WM8805 is run in master mode, so as to minimize jitter due to the RPi’s problematic I2S clocking scheme. The WM8805 is clocked by the same oscillators that are used to reclock the s/pdif signal.
The WM8805’s s/pdif output goes through an Si8641 150MHz galvanic isolator and is passed to the “clean” side of the board.
There the signal is reclocked by a high quality flip-flop clocked by high quality NDK oscillators (housed inside a metal box, used for shielding against EMI/RFI). There exist two oscillators, one for the 44.1K family and one for the 48K family of sampling rates. The output of the oscillators is put through NB3L553s for buffering and isolation.
The entire isolated part of the board is powered by a DC-to-DC converter that offers galvanic isolation. Following this converter there exist a large number of LDO regulators and filter components. An LT3042 regulator is used to power one of the most critical parts of the circuit: the flip-flops that do the final reclocking.
The answer is, surprisingly well for the money.
My RPi stack included an RPi 3 with the DigiOne, powered by Salas’ new L-Adapter power supply and running Archphile. The music was coming from my NAS. No audiophile ethernet switches were employed. 😛
Pitted against that I had my Logitech Squeezebox Touch running the EDO plugin for up to 192K s/pdif from its coax output and my relatively pricey Pioneer DV-LX50 Universal Player (using its coax s/pdif output).
The music used was Dire Straits’ SACD album (having selected its CD (and not SACD) layer) which was also accurately ripped to my NAS.
Output from the s/pdif transports went into an AK4118-based s/pdif receiver of my own design which in turn feeds my dual mono AK4493 DAC. The DAC’s output goes through a Salas DCG3 preamp into my Hypex amp.
First up was the Pioneer. It had been a while since I had listened to it through its s/pdif output so I was in for a bit of a shock. Its output sounded coarse, strained, tiring. For a moment I thought that it was due to the SACD’s mastering (the CD layers of SACDs are rumored to be mastered intentionally bad so as to give the impression that the SACD layers sound even better than they actually do), but that changed when I switched to the Squeezebox. Things got noticeably better, actually listenable. Not exactly close to what I had been accustomed to using the Squeezebox’s USB port, but closer.
Then I switched to the DigiOne. Wow! All of the “coldness” of the music was gone, the stage gained depth and width, the music became more detailed and lifelike. This was definitely a step up.
I would dare say that this s/pdif setup came in fact close in SQ to my USB setup. This was a very pleasant surprise.
Now I need to do some A-B testing between the DigiOne and the USB output of the RPi. So to-be-continued..
About a month ago the Raspberry Pi 4 was announced, pretty much blind-sighting everybody.
For the last (many) years, since the announcement of the RPi 2, we had been used to relatively minor incremental upgrades every time a new RPi came out.
Usually the new processor was a bit faster, we got WiFi and BT, then better WiFi, then faster (almost) GbE network, etc. But until now, all of these connectivity options had to be accommodated by a single USB 2.0 port on the SoC.
But this year everything changed. We got a new SoC (the BCM2711), one that finally supported an RGMII interface for a true GbE port, plus a PCI Express port that is used to give USB 3.0 & 2.0 connectivity at useful speeds.
We also got more processing speed and more RAM options, up to 4GB of fast LPDDR4 memory, dual HDMI outputs, etc.
So, all of the above specs mean that the RPi is definitely faster and more capable than ever as a desktop replacement. But is it indeed a better audio streamer for us audiophiles?
For starters, it’s been almost a month since its announcement and availability (I got my unit delivered just 3 days after its announcement) and AFAIK the well-known audio distributions do not yet support it.
Then there is the increased system complexity and power consumption that comes with the new architecture. More power consumption and more ICs usually mean more noise. More noise is never good news for audio.
So I had to do some testing. The idea was to compare the RPi 3 that I had for a couple of years now to the RPi 4.
To keep the playing field as level as possible both of them were running the exact same software (Raspbian Buster Lite, since ATM that is pretty much the only OS that supports both of the platforms) with MPD loaded and were powered by the same (excellent) Salas L-Adapter power supply.
Connection to my DAC (DIY dual AK4493, very detailed) was through USB 2.0.
The music streamed from a NAS box over Ethernet.
I had a friend over in order to at least try to have a bigger sample size (of ears).
The music used was a handful of tracks that we always use for such comparisons (well known material).
We listened using the RPi3, then shut it down and booted up the RPi4, listening to the same material.
Much to our surprise, we actually preferred the sound of the RPi3!
The RPi4’s presentation had something of a “fatiguing” effect. The sound was a bit more “coarse” that that of the RPi3.
We are not talking about big differences here, but they were there. Note that my system is pretty resolving, every change to any component is audible, so YMMV.
I’m not saying that my (our) results are 100% conclusive, but in any case it seems like I’ll be going ahead with my “Audio Pi” project after all (I was considering waiting for the Compute Module 4 to come out).
Not much free time these days so updates have been slow.. but I have a lot of interesting stuff cooking in the back burner.
One of them is an audio grade RPi.
Essentially it will be a Compute Module 3 on a mainboard loaded with ultra low noise linear power supplies and some necessary peripherals.
The idea came to me quite some time ago but it wasn’t until last November that I decided to actually go ahead with it.
The proof-of-concept PCBs for the mainboard were done by December.
It appears that even the PoC board, with average quality power supplies, has a cleaner I2S output compared to a standard RPi3 powered by an equivalent linear power supply:
The next part was the PoC board for the USB Hub & Ethernet controller. That took a bit more time and a 4-layer PCB with numerous 0402 sized components but it too ended up just fine (with the exception of a bad RJ45 footprint..).
So now I have a fully functioning set of boards with average quality power supplies that already performs better than my Squeezebox Touch as a USB transport.
Next step is to design a single board integrating all of the components plus ultra high quality power supplies.
That will probably be a summer project..
I apologize in advance if this sounds a bit elitist on my part, but it amazes me just how many DIY audio hobbyists need help planning their power supply solutions.
And I’m not talking about “Salas or LPS-1” discussions, I’m talking about “what voltage should my transformer put out” type of discussions.
So I’ll attempt to clear up the basics.
First up, you’ll need to know your load. This means voltage and maximum current requirements. Based on those, you will know what your options on power supplies will be. The options are too many to get into – switching vs. linear, series vs. shunt, LDOs vs. batteries, etc etc.
The point of this post is not to help you select a power supply topology – that decision is largely subjective anyway.
For our exercise, we’ll assume that our load requires a power supply capable of outputting 5V at up to 1A. We’ll also assume that we’ll be making a “classic” LM317 regulator based power supply.
This is the part of the power supply’s circuit that we’ll be focusing on.
Our task is to select a proper power transformer, rectification stage and filtering capacitor.
Selecting the transformer’s output voltage
Looking at the LM317’s datasheet we see that it requires an input-to-output voltage difference of at least 3V to function properly. This means that in order to get regulated 5V at its output, its input unregulated voltage will need to be at least 8V.
To get 8VDC after rectification and filtering, our power supply will need to supply at least 7V AC. This 7V AC will become 7 x 1.414 = 9.9V – 1.8V (worst case voltage drop on the rectifier diodes) = 8.1V DC.
In real life we will need to take into account possible “sagging” of the power grid by a few volts during certain hours of the day, so it would be a good idea to compensate for that by choosing a transformer with an output voltage slightly higher than the theoretical one. In our case, instead of 7VAC a safer choice would be ~8VAC.
Selecting the transformer’s power rating
Power rating in transformers is expressed as “Volt Ampers” (VA), also known as Watts. It’s the product of the transformer’s output voltage times its rated output current. So a transformer that is characterized as “12V 120VA” is capable of outputting 12VAC at 10A.
Going back to our example, we’ve stated that our load requires 5V at 1A. We have already calculated our transformer’s necessary output voltage to be 8VAC, so 8 VAC times 1A equals 8VA, right? I’m afraid not. This is the most common pitfall for electronics hobbyists when it comes to power supplies. They assume that a transformer rated at 12VAC @ 10A can in fact still deliver 10A after the voltage has been rectified and filtered. But that can not happen. If it did, it would mean that the transformer is outputting more power than what is being put into it.
The thing is, the total power that can be “transformed” by a transformer is fixed, so since the rectification and filtering results in a DC voltage higher than the available AC voltage, the corresponding maximum current must be smaller.
So in our case, to get 1A DC out of our 8VAC transformer we will need a transformer rated for at least 1.5A of current, so 8 x 1.5 = 12VA.
Realistically, to have the transformer running cool and noise-free, we’d double that and go for about 25VA.
Component selection: Rectifier diodes
In order to keep the diodes running cool and reliably you should choose parts rated for at least three or more times your expected load current. This is especially important when building shunt power supplies which draw constantly relatively high currents. For audio circuits opt for ultra fast recovery diodes, such as the MUR series (~25ns). In case of high currents (>1A) be sure to either mount the diodes at least a few mm off the board (in case of radial parts) or use heat sinks (in case of diodes that can accept them). Each diode drops up to about 0.9V, so when they are passing ~1A of current they will need to dissipate almost 1W of heat. That is quite a lot of heat for a small part.
Component selection: Filter capacitor(s)
The filter capacitor is a hard-working component. It has to charge and discharge about 100 times a second (120 times in the US), so as to smooth out the fluctuating voltage that comes out of the rectifying stage. The more current the load is pulling, the harder the capacitor has to work. In our example, since the capacitor is quite possibly charging to the maximum available voltage coming out of the rectifiers, its voltage rating has to be at least (8VAC x 1.414) – 1.8 = 9.5VDC. Taking into account the fact that transformers under no load output a voltage that is higher than their rated voltage, you should go for a capacitor with a voltage rating that is reasonably higher than the minimum required.
Regarding the value of the capacitor, things are just a bit more difficult to figure out, but in the end it all boils down to one thing: how much voltage ripple are we OK with (a.k.a.: is our regulator and load able to tolerate). Once we have determined that, all we need to do is do the math. An excellent description of the theory behind this is this page: http://www.skillbank.co.uk/psu/smoothing.htm
So, C = Iload / 4 * f * Vpk-pk ripple. For example, let’s say that we would be OK with 0.5V ripple. We have: C = 1A / 4 * 50 * 0.5 = 1A / 100 = 0.01F = 10.000uF
One last thing you should keep in mind is the ripple current rating of the capacitor. A good guesstimate is a value at least two times your expected load consumption, but that may vary a lot when you get into high performance audio grade power supplies. For more information (that is beyond the scope of this post) have a look here: http://www.skillbank.co.uk/psu/ripple.htm The spreadsheet linked at the bottom of the page is a great resource.
It has come to my attention that the “Contact me” form add-on I’ve been using was kind of buggy. The kind of buggy that apparently led to me receiving approximately half of the messages sent to me.
This has been going on for (most likely) about a year and has been fixed only lately (since about a month ago).
So, if you’ve tried to contact me via this form in the past but have received no reply, it was due to this bug. I definitely reply to everyone that contacts me, even though at times it might take me quite some time.
So, anyone who’s tried to contact me with no success, please try again.