I got these photos almost a year ago and they were “lost” in my inbox.
This one uses a yellow LCD. Actually pretty good looking. The quality and finish is of these diy projects is amazing!
The ATC’s (speakers) are custom 150 ASL anniversaries with discrete active amp packs but the room is temporary and centre is coming off the floor asap but it currently sounds nothing short of awesome with the Buff2’s driving the amps direct using legato 3.1.
The PC can decode and output blu ray and full DTS MA HD multi-channel thanks to J River Media Centre which is a superb piece of software.
Getting multi-channel remote volume control without compromising quality was a problem until your code. Next challenge for me will be more buffalo to use the pc for FIR active crossover for the ATC’s…
Dual Mono Buffalo II
The Arduino is well positioned having the USB connector easily accessible for firmware upgrade.
You can read part I here: [link]
You can also read another build log here: [link]
VERSIONS OF DIYINHK BOARDS
I have been able to identify 5 different versions. My board is V2. Each version has a new input pin configuration and/or other enhancements.
V1: Hardwired for I2S plus additional SPDIF iputs
V2 (my board). Access to all the inputs (must cut the shorting trace for I2S), more chip bypass.
V4. Introduces a contiguous ground plane (big deal) and other enhancements:
Another V4 board [link]:
V5. Separates AVCC-L and AVCC-R:
SEPARATING AVCC-R AND AVCC-L (IN EARLIER VERSIONS OF THE BOARD)
Easy mod to separate the power connections for AVCC for the right side and the left side.
The board neatly connects the left and right side AVCC pins as shown in this photo [link] of V4 of the board, showing the main trace for AVCC that is under the DAC chip. On the other side of the board are bridges from the main trace to the individual AVCC pins. Cutting those bridges effectively separates AVCC-R from AVCC-L.
Here is a photo of V2. It seems that the bridge pattern matches the AVCC trace of the V4 board. I’ve measured the connections of all the bridges and they connect to AVCC.
Here is an overlay of the V2 board traces on the V4 front side photo. Seems like a workable solution. You can also see that each of the 8 AVCC pins has its own bypass capacitor. Hmmm, since there are 8 internal DACs in this chip, this means each internal DAC has its own supply pin.
Connecting the AVCC-R traces together:
And cut the bridges:
Measurements confirm that the AVCC-R and AVCC-L are no longer connected. This mod should work at least in V2 and V4 boards.
Separately powering AVCC-L and AVCC-R has audible advantages as reported by diyinhk [link]:
Separating the ES9018 AVCC_R/L power supplies provide a much more 3D sound stage. The location of every instrumentals is much much more clear
It’s like listening the differences between an upright piano and a grand piano.
This is really a breakthrough even after many layout improvements since the first version. I finally know why many diy’ers want dual mono config or at least a seperate AVCC_R/L power supply.
One option is to use TPA’s AVCC which is a dual shunt regulator board. A diyer just did that [link]:
Since I updated the AVCC board in my BII DAC [link], I also have the older version to use.
ESS OPAMP BUFFER
Another option is to use the ESS opamp buffer as specified in the ESS documentation [link].
ESS specifies a single buffer for both channels, but this was for their original ES9018 evaluation board with a 40 MHz clock and 3.3V supply. If one uses a faster clock and increase the operating voltage a bit above the 3.3V, then the power requirement would be higher as shown in this graph:
Notice that if the clock frequency is 100 MHz, then the current requirement for AVCC at 3,3V is beyond 50 mA. Most opamps max out around 50 mA, thus using two opamp buffers makes a lot sense.
After upgrading my Buffalo II with the AVCC 2.1, the original AVCC v1 became available. No reason not to reuse it.
Removing AVCC module from BII: It wasn’t easy to desolder it from the BII (I had soldered the module to the pins rather than using the pin header for plug-in). Luckily the two “IN” connectors are tied together so connecting the input on one side is sufficient.
With cat-5 cables making 4 connectors, it can fit on the board connecting GND and AVCC L and AVCC R
Connecting to AVCC L pin and AVCC R pin (after separating AVCC) and ground pads
The input to AVCC needs to be 5V (5.5V is maximum for this version of the AVCC board). It can be fed externally without connecting FB3. This way the AVCC/2 offset voltage used by the opamps can be generated without any modifications. For ultra-accuracy, the AVCC/2 offset voltage can be separated between L and R by using a similar resistor divider (externally implemented) for the other channel and by cutting the AVCC/2 offset line between L and R channels
I also tested it to ensure good working condition:
It is designed for 3.5V output. The output would settle to about 3.51V because the reference voltage is based on an LED and LEDs have a negative temperature coefficient, so as it warms up, its forward voltage decreases slightly.
This module has “uneven” LED brightness. It was like this since day one [link], but TPA assured me that it is of no consequence. I tested it with a 70 ohm load (drawing 50 mA per side) and measure the output with my rudimentary mini scope. The output looks clean.
For $25 I purchased two SURE Electronics TPA3122D2-based amp modules from eBay (link).
It is an extremely simple configuration: input DC blocking capacitor, output low pass filter and power, and a few control lines. I suspect sound quality is overwhelmingly determined by the chip and not so much by implementation. Of course some minimum level of quality is required for all the components. The kit is also suitable for beginners because all the components are through-hole and there is ample space.
There is a discussion on these Ti TPA31xx amps over at diyaudio [link]. The reports are good.
I plan on using them in BTL configuration (balanced input) with the $99 ES9018 DAC in voltage mode out. With Hifiduino S/W control, there is no need for external volume control resulting in a compact and “straight-through” DAC-AMP configuration. 20~30W per channel is plenty power for the Alpair 12p.
Here are some photos:
Very nice quality board. (Nowadays almost any PCB from China is top-notch in quality). Just the board itself is worth the price of the kit
Hmmm, not a solid ground plane. May need to add some bridges for the ground current as I did with the $99 ES9018 board. Easy to do mod if needed…
For the power ground currents, there is nothing to do; there is a straight path to the power ground connection in the board.
For the signal ground currents,
Here is the circuit diagram of the board implementation
Here is the circuit diagram of the TPA3122D2 Evaluation Board [link]
And here is a photo of a stock implementation (courtesy of PartsExpress -the Amp is also available from PE)
The plastic storage case for the components is actually pretty nice (at least worth $.99 at the local 99cents store )
No-name components… replace with your favorite parts
The output inductor is pretty beefy. Should work pretty well if not that the value is listed as 10 uH (For BTL into 8 ohm, it requires 22 uF according to the data sheet. For a replacement, people at diyaudio have recommended the WURTH inductors (not specifically for this amp, but other class D) such as the Wurth WE-PD (sheilded) series inductors [link] at $3.66 each. Toroidal inductors are also a good choice because the magnetic field is confined within the toroid. This seems a good candidate at at $2 each: [link]
The TPA3122D2, current version of the chip [link]
Have some film-foil caps I plan to use for the input DC blocking capacitor (Multicap PPFX). These are an improvement to the including capacitors in the kit. (These are pricy caps, but they have been sitting in my drawer for years )
Other low cost are:
Film and Foil
- Dayton film and foil: [link] Parallel two .47uF capacitors to get the desired value. These are the most cost effective film and foil capacitors.
The amp has better performance if configured as mono, balanced input (BTL mode). Notice that at 20 W output, the BTL mode outperforms the SE at 1 W output:
Here is comparing with a newest generation TPA3116D2. All of these chips have similar performance, the newer chips have better numbers at the frequency extremes…
PRETTY GOOD PERFORMANCE
If we compare the two devices (5W trace), we get the following plot. The older TPA3122D2 is comparable to the latest devices. One thing to note is that the spec sheet uses a gain of 20 db for the 3122 (and 3123) and a gain of 26 db for the 3116 part. If we use the same gain for both devices, the 3116 will perform markedly better.
OUTPUT DC BLOCKING CAPACITOR
There is another advantage of of configuring this Amp in BTL: the output DC-blocking capacitor is not needed.
The circuit is built to support SE configuration. As such there is space for two 220uF electrolytic capacitors in series with the speaker output leads. If configured for BTL, these capacitors which can negatively impact the sound, can be omitted.
The circuit diagram for BTL configuration is shown here (taken from the datasheet). Notice how simple is the circuit: input DC blocking capacitor, power filter and bypass caps, output low pass LC filter. The few other capacitors are clearly explained in the datasheet.
Input and bypass caps
Put high quality film caps on LIN, RIN, and BYPASS (C1, C2, C14). This will be the biggest factor in improving sound quality.
[Rule of thumb -- make the BYPASS cap value the same as the input cap value. The input cap value should be based on your required bass cut off, which is determined by the input impedance, which is controlled by the amp gain. The minimal input impedance at the 36dB gain selection is 9Kohm, while the maximum impedance is 60Kohm at 20dB gain (all this is in the TPA3122D2 data sheet). Use the standard filter formula to determine this value: C = 1 / (2 * 3.14 * frequency * impedance)]
The two input caps in the BOM appear to be mylar film caps. Replacing these with good polypropylene caps will improve the sound quality. The BYPASS cap (C14) is a mono ceramic. The BYPASS pin is the feedback loop bypass of the analog first stage. It definitely needs a quality polypropylene cap. I’ve found the change in audio quality to be noticeable.
Analog supply isolation
Replace C22 with a larger value electrolytic (100uF or so) and a small resistor (100-220 ohms) between VCC and C22. This will isolate the internal opamp stage from the switching stage.
The analog stage power supply isolation mod is a more subtle mod. The internal analog stage opamps theoretically should cancel out 100% of the power supply noise via common mode rejection. I’ve found this to be rarely the case. Further, the majority of the power supply noise is coming from the switching frequency of the digital output stage, which means the input stage opamps are doing the greatest amount of feedback at the output stage’s switching frequency. If you think about it, the analog stage is doing the adjustment right at the point the digital stage is switching — it’s not a clean path for an audio signal. Adding a filter for the analog stage power supply reduces this potential for distortion.
Power supply capacitors
For more bass, replace the C7/C8 and C10/C11 pairs with a 2200 uF (8 ohm speaker) or 4700 uF (4 ohm speaker) and make sure the power supply you plug into the board has at least double that value.
POWER SUPPLY CAPACITORS MOD
The datasheet recommends the use of low ESR capacitors. A good candidate is the new Panasonic FR, which is the successor of the popular FM. The board can fit capacitors of 3.5 mm lead spacing and 8 mm in diameter. There are space for 4 PS capacitors. Since we are not using the output DC blocking capacitors, we can potentially use this space to fit more capacitors (mod the connections) or larger capacitors
Panasonic is introducing the FR-Series, new Aluminium Electrolytic Capacitors in Radial Construction. This capacitor is the perfect solution for applications, which require ultra low ESR – very high ripple current – very long life in a small mounting form.
In comparison to our current products e.g. FM series, we could achieve a lifetime upgrade of up to 100% (up to 10000h at 105°C) and a capacitance increase of up to 30% by improving the material technology.
The largest 35V 3.5mm lead spacing capacitor is 470 uF [link]
A simple configuration is to use a single 2200 uF capacitor (or two, one above and one below) and install it horizontally.
POWER BYPASS CAPACITOR MOD
Solder .1uF mono caps right across Pin1-Pin20, Pin10-Pin11, and Pin7/8-Pin16/17. This will put these bypass capacitors as close to the chip as possible
POWER CONNECTION MODS
Input protection diode
The power input line has a diode for reverse-polarity protection. The evaluation board does not have such diode. I think I will omit the diode and just put a jumper.
The power line snakes around from one side to the other. There is opportunity to shorten these lines and make the supply lines in a more symmetric manner.
ANALOG SUPPLY ISOLATION MOD
The analog supply is tied to the digital supply. Internally, there is a regulator providing a regulated (and lower) voltage to the analog section as shown in the block diagram:
It is simple to isolate this supply from the digital supply. The following are potential methods:
- RC filter between VCC and AVCC. The resistor will drop some voltage, but full voltage is not requires as there is an internal voltage regulator
- Pi filter CLR or CRC for improved filtering
OUTPUT FILTER INDUCTOR MOD
As indicated above, the output inductor needs to be changed to 22 uF for BTL mode into 8 ohm according to the spec sheet. According to “Design Considerations for Class D Amps” [link]
The output inductors are the key elements in the performance of the class-D audio power amplifier system. The most important specifications for the inductor are the dc resistance and the dc and peak current ratings. The dc resistance directly impacts the efficiency by adding to the total load resistance seen by the power supply.
The inductor current ratings must be high enough to avoid magnetic saturation, which will cause an increase in audio signal distortion or, if completely saturated, will cause the inductor to appear as a short rather than an open circuit to the PWM output. This could potentially damage the device or speakers from the resulting high current surge that may occur during turn on, or the increased quiescent current during normal operation. It would seem best, then, to choose an inductor that has a much higher current rating. The tradeoff is that the size and cost increase as the current capability increases. Shielded inductors will also help reduce distortion and EMI, minimizing crosstalk in the process.
Of the two inductor options considered above:
- Wurth WE-PD (sheilded) series inductors [link] at $3.66 each.
- Bourns 2100 series inductor [link] at $2 each
The Bourns will perform better (according to the TI document) due to its lower resistance (15 vs 43 mOhm) and higher current rating (7 vs 4.1 Amp), plus it is cheaper!
OUTPUT FILTER CAPACITOR MOD
According to “Design Considerations for Class D Amps” [link]
The filter capacitors should be ceramic capacitors with X7R characteristics for stability over voltage and temperature, and can be found in common
surface-mount packages as small as 0805. The values of capacitance calculated in the example above are readily available in ceramic chip and metal film
capacitor product lines. Measurements have shown little difference between the performance of these two types of capacitors, though some audiophiles will strongly recommend the metal film.
ADDITIONAL WISDOM FROM THE FORUMS
The only difference between the various TI TPA31xx class-D medium power chips is their thermal rating. They are actually all the same piece of silicon, just in different packages. You can glue a heatsink to any of them and get the same thermal derating.
The TPA3122D2 can take a recommended maximum VCC of 30V and will put out ~47W into a bridged 8 ohm load. It has built in thermal protection (rated 150C trip point), so you couldn’t overheat it if you wanted to.
A really easy heat sink is to mount the DIP chip underneath your PCB with all the rest of the components on top, then mount the PCB on a chassis so the DIP package contacts it (an aluminum plate in a project box will do nicely).
Your 10% THD figures are totally dependent on VCC. As long as the peak output doesn’t saturate the rails, the distortion figures will remain well below 1%. If we assume a 90% efficiency, then 12V VCC gives 10.8V P-P across a load. That’s approximately 3.6W peak/2.6W RMS for 8 ohms single ended mode. At 24V VCC, you get 21.6V P-P and approx. 14.6W peak/10.3W RMS. If you look at the TPA3122D2 data sheet, figure 6., you’ll see that the 10% distortion figure pretty much matches these calculations. Of course the efficiency level changes with power output (figure 14.), so you’d have to take that into account when making final calculations.
Even at peak power in BTL mode, these chips won’t dissipate more than a few watts, so I wouldn’t worry too much about heatsinking.
The switching frequency of the TPA3112D1 is at 310KHz median vs. the TPA3122D2 250KHz median which isn’t a whole lot of difference. The use of bead filters is merely an economical trade off for low cost applications. If you want good filtering with proper loading, you’re going to have to go with a standard LC filter.
Effect of output filter: [link]
Discussion link: diyaudio
Design link: Audio Design Guide
This year brought two brand new implementations of the Sabre32 ES9018 DAC to the diy community. One is the bare-bones $99 ES9018 board from diyinhk and the other is a fully assembled board from Quang Hao-Andrea, the “DAC-END R”. There is a group-buy going on here: [link].
Joining the Buffalo DAC and the Acko DAC, there is now a version for every taste, budget and diy capability. It’s never been a better time for the diyer.
The DAC-END-R ES9018 is fully assembled including the power supplies. All you need to add is a transformer and the input connections. There is even a “plug & play” header and mounting holes for the Amanero USB/DSD interface. In effect, this is not quite “diy” as it can be ordered fully functional with power supply and enclosure. But it is also not a commercial offering since it is currently a “diy group-buy” project.
I was able to get a hold of a prototype board to test it out. As can be seen the photos, the finish and workmanship is a combination of machine reflow and manual soldering. Quang Hao indicates that the production board will be totally machine manufactured for a fully professional finish.
The all popular Crystek CCHD-950, 100 MHz. Using this speed, the max specified in the datasheet allows noise-free playback of 384KHz sample rate material.
The board has built-in passive I/V resistors (the production board will have MELF resistors instead of the Caddock resistors shown here). The small 10 ohm allows the DAC to work “closer” to current mode (making the voltage swing smaller) improving its THD performance. Here is a post explaining the value of small value passive I/V: [link]
The DAC can operate in both current and voltage modes. With either mode, you can also bypass these resistors by removing the jumpers (e.g., J8) and use the DAC in proper current mode (fixing its voltage) or in full voltage mode (allowing full voltage swing) for direct connection to an amplifier.
Here is a photo of a production board
The DAC board comes with nine power supplies:
Two Pre-regulators based on the LT1963 high-power low noise LDOs. These regulate the rectified ~9V DC from the transformers and provide the “raw” 5V for the analog sections and the “raw” 5V for the digital sections
Five shunt regulators based on the Ti/BB OPA2134 opamp:
- Two shunt regulators for the 3.3V Analog R and L
- Two shunt regulators for the 1.2V Analog R and L
- The fifth shunt provides the 3.3V digital for the clock
The core 1.2 digital supply is based on the LT1963 low noise LDO.
There is also an additional digital 3.3v (yet another LT1963 in a different package) regulator for the controller board and LCD display and also as a second pre-regulator to the 1.2V digital supply
I can tell Quang Hao is a fan of the LT1963 regulator. Every package variation has been implemented in this board
All the regulators can be bypassed with the built-in jumpers (in case you wish to try other regulators).
Included with the DAC board is a controller board with LCD display and infrared remote sensor. This is aimed at the “non-tinkerer” (unlike my Hifiduino code )
There are 5 buttons with the following functions
- Increase volume (up button)
- Decrease volume (down button)
- Mute (center button)
- Switch input forward (right button)
- Switch input reverse (left button)
The LCD displays volume level, input selection, and the sample rate when there is lock on a signal.
On the left of the LCD is the IR sensor. I have not purchased the remote (widely available in eBay) so I have not used the remote functions. I suppose it does the same as the buttons.
The board has its own power supply if you do not wish to power it with the 3.3V supply in the DAC board (there is a jumper to disable powering the control board).
SMART INPUT CONFIGURATION
I am happy that Quang Hao/Andrea decided to follow my post about leveraging the automatic detection and internal mux capability of the Sabre DAC [link]. As such, the DAC can switch inputs by just programming the internal registers without the need of any external circuitry. This makes switching inputs “totally transparent” without having the signal go through relays, additional wiring or switches.
I tested this configuration with an optical SPDIF connection and with the Amanero I2S/DSD connection both connected to the DAC board. The SPDIF and the Amanero board had their signals live at all times (meaning that if I switch to either one, there was a lock to the signal). The source switching worked flawlessly. And no problems were observed. There was some concerns about this not fully working, but as implemented and tested, it works perfectly.
The DAC can accommodate two additional spdif inputs: a coax and an AES/EBU each with the appropriate isolation transformer and level converter as shown below:
This is a very nice implementation of “smart” input selection.
Voltage-mode output gh the same serial interface (don’t have any DSD higher than 128). Every sample rate and format was supported without a glitch (Some may recall that with the 80Mhz clock, playing 352K and 384K had noise glitches)
All in all, I felt the DAC-END R implementation excels in the top end. Sometimes I felt a more detailed presentation in the top end as compared to my memory on familiar tracks. No doubt the use of the faster clock and the use of shunt regulators throughout had something to do with it.
Here are some photos of the production boards for the diyaudio group-buy I received from Mr QuangHao
Notice that the oscillator has been replaced with a different one:
According to QuangHao, Mouser did not deliver the CCHD-950 he ordered, so he had to purchase CCHD-575 [link]
Comparing the phase noise between the two shows that the CCHD-575 is actually slightly better than the CCHD-950 in terms of phase jitter:
Here are some photos I pulled from diyaudio. Looking very nice!
Very clean design. A single custom transformer provides the 4 required voltages.
The display is now a nicer blue/white LCD…
GROUND PLANE MOD
Note: the current version of the board has a solid ground plane, so this mod only applies to early versions (like mine) of the board. You can see the new version here: [link]
It is an industry best practice to have a continuous and mostly uninterrupted ground plane in any layout “no matter what”. This allows for uninterrupted ground current returns, following a lowest impedance path and also terminating (confining) the electric fields of power and signal lines. Now this doesn’t mean “completely solid”, but the aim is to have as continuous as possible. In a two-layer approach, this is harder, but according to industry documentation, a well designed 2-layer approach would approach 98% of the capability of a 4-layer PCB.
The return ground current likes to flow under or along the signal paths. That is, if you have electrons flowing in the signal paths on one direction, then there are electrons flowing in the opposite direction right under or along the signal lines. This is just the law of physics, otherwise you will be piling up electrons on one side of a conductor and this is impossible. (It is more complex than this, but this is the simplest explanation).
According to “Op Amp Applications Handbook” by Walt Jung, p. 640 (link),
Whenever there is a break in the ground plane beneath a conductor, the ground plane return current must by necessity flow around the break. As a result both the inductance and the vulnerability of the circuit to external fields are increased. This situation is diagrammed in figure 7-32 where conductors A and B must cross one another
This board has mostly a continuous ground plane except it is broken by the opamp power lines. If you follow the output of the DAC, there are 4 lines that connect DAC to opamps. The return ground current under the signal path would want to follow the signal path but it is interrupted by the power lines and thus it has to flow around. This is cause for added distortion (how much and whether audible or not? I don’t know).
In order to remedy this, I’ve installed bridges right under the signal path where it is interrupted by the power lines. In fact there are 5 such locations in the board.
The 5th location is this one. There is no “6th location” because in that location, the ground plane is uninterrupted.
I think, 24 gauge (cat-5 cable) wire is sufficient. Here is compared with an 18 gauge cable. Flattening the wire allows for easier soldering (it won’t roll around the board). Actually I will install a wider “strip”. See the text below.
Scrapping off the pads takes some patience and for me, a lighted magnifying lens
According to this document from TI [link]
One thing that many people forget about is for a current to flow out to a point, there MUST be a return path or else current will Not flow. Since there is a current flow, then the return current flow will find a way back to its’ source one way or another.
Return current density is highest directly under (or over) the signal trace it was sourced from. Even if a solid ground plane is used, the concentration of current flow will still be adjacent to the signal source trace.
Therefore, a wider bridge is better than a narrow bridge, but certainly a completely solid ground plane (which is desired) is not required to handle the return current.
According to “Successful PC Grounding” the return current path has been characterize with relation to the signal frequency. At low frequencies (1KHz), the return current mostly flows across the shortest physical distance. At high frequencies (1MHz), the return current path is mainly under the signal path. This basically says that the return current follows the path of least inductance.
The following two diagrams shows the return current path for low and high frequency signals red/yellow/green showing the highest current concentration.
This means that at audio frequencies, a large portion of the ground return current will follow a path of shortest physical distance.
I’ve added another path for the signal ground return current to flow back to the DAC, one straight from the ground pins of the single ended output to the DAC chip, crossing the power lines.
Here are the photos with the installed ground current bridges:
I suspect this is the path of the return ground current. As there are no ground connections in any of the opamps, the signal return current is that which is coming back from the components downstream from the DAC. At audio frequencies, most of the return current would want to follow a path of shortest physical distance. Some will flow back under the signal lines.
It can be noted that even without the return current bridges, the return current flow with the original board is not that bad: the current just makes a long curve on its way to the DAC.
The new version of the board has a solid ground plane
Tried to figure out what’s the best way to install a removable clock on this board. Ian had developed a clock carrier board, so I decided to use it. Here is what I have:
Installed a female pin header/socket connector to the carrier board
Used smaller headers that “standard”. These are metric 2 mm pitch headers. The typical ones are 0.1″ pitch. The pins are the ones used to connect the AVCC to the Buffalo DAC
Scraped the solder mask to expose the Vcc and GND planes. Since the connector is low profile, the clock can also be used in Ian’s FIFO reclocker board by installing the pins. Clock out and enable will be connected with wires to the respective pads.
The clock board connects to pins that are installed in the clock pads in the DAC board in the following manner. I bent the pins first, put them on the pin header and soldered them to the pads -an easy job.
The clock carrier board connected to the DAC board:
SAW (Surface Acoustic Wave) clocks
There has been some interest in using SAW clocks for the ES9018 DAC, and favorable results have been reported [link] even though these oscillators measure poorly as compared with the Crystek clocks typically used with the ES9018:
A 100mHz Epson SAW was compared to the 100mHz Crystek fitted as standard to the Twisted Pear BII… The outputs of each dac could be switched on the fly to feed the pair of I/V transformers. Of the 5 listeners, 3 clearly favored the SAW, and 2 (including myself) were undecided though one of us would choose the Crystek if forced to. The SAW, even to me, appears to have better resolution.
According the Epson, the SAW oscillators have advantage over traditional crystal oscillators. I’ve pulled this info from their documentation
|SAW||Fundamental, high frequency oscillation and high drive||Epson XG-1000CA (106 Mhz)||Excellent. The noise floor is a low because of the high drive operation, and the high frequency is a steady fundamental oscillation.|
|Crystal (Fundamental)||Fundamental, high frequency oscillation and low drive||Excellent. Q value of the AT crystal unit is high and a fundamental oscillation though the noise floor goes up more than the SAW oscillator due to the lower drive level.|
|Crystal (3rd Overtone)||Overtone, high frequency oscillation, low drive and fundamental supression||Crystek CCHD-950 (100 MHz)||Good. The oscillation stability is inferior to two above-mentioned methods because the fundamental oscillation is suppressed and overtone is oscillated.|
Note: 3.3v 100 MHz parts are not available from Digikey, so I linked 106 MHz parts for reference. It is not recommended to use parts in excess of 100 MHz
Another good part are the FOX Xpresso oscillators. Available for under $4 for a 100 MHz part, 25 ppm frequency stability, 3.3V and in 7×5 mm size [link]
Jitter for a 100 MHz part can be calculated from the measurement plot provided in the datasheet (I have plotted the curve for the Crystek CCHD-80 which was standard equipment on the original Buffalo II DAC.
The phase jitter value 10Hz-1MHz for a 100Mhz part is approx a very respectable 2.7 psec. RMS.
ES9018 POWER CONSUMPTION [link]
Analog 3.3V supply
- AVCC-R (3.3V) = 32mA
- ACCC-L (3.3V) = 32mA
Buffalo II AVCC is 50 mA per side including the shunt current through the regulator [link]
Current consumption for AVCC also depends on the clock frequency and the actual voltage. The chip can operate all the way to 4 V (although many do not recommend doing so). According to a “Analog Power Supply Consumption” [link], the relation with voltage and frequency is as follows (I took the data from ESS and extrapolated it to 100 MHz and also beyond 3.8V):
- AVDDL (1.2V)= 8mA
- AVDDR (1.2V)= 8mA
Digital 1.2V supply. This is the “core” of the DAC
- DVDD (1.2V) = 105mA
KlipschKid over at diyaudio [link] has done some measurements of the 1.2V supply (which combines both the analog and digital 1.2v supplies) with respect to clock frequency.
I’ve plotted his results (subtracting 5 mA used by the 7805 regulator)
There is also the current value for a 50 MHz SAW oscillator, at 110 mA [link] which is kind of on the high side but still ballpark value.
I had previously measured [link] the power consumption for the B-II 80 MHz DAC and found the current to be at 250 mA total. If we subtract the 100 mA for the AVCC and the 10 mA for the clock, we end up with 140 mA for the 1.2V(+3.3V digital) supply. This matches well with the graph above.
If playing 192KHz sample rate vs 44.1KHz, the overall current consumption increases 30 mA.
Digital 3.3V supply
- DVCC_T (3.3V) = 5mA
- DVCC_B (3.3V)= 5mA
- VOSC (3.3V)= 10mA. Actually for the CCHD-957/950, the current consumption of the oscillator is 15 mA typical and 25 mA max
IMPLEMENTING THE 1.2V SUPPLY
There are three 1.2V supplies for the ES9018 chip, analog left, analog right and digital core. these are supplied by a single 1.2V regulator.
Here is some evidence that there is no need to feed the analog supplies separate from the digital supply (NOTE: most of my knowledge on this comes from the good people at diyaudio especially Russ of TPA who have shared a lot of information with us diy folks.):
I conversed with Dustin about this at length as well as experimented on my own. The 1.2V supplies that are on the analog side are actually just driving level shifter gates into the modulators from the core. It makes no difference if they are powered separately whatsoever. [link], [link]
The 1.2V supplies only do two things in the DAC, the first is that it drives the core of the chip. This is of course crucial. The second is it drives the gates of level shifters (a high impedance) into the quantizers.
It is important to understand what these level shifters do. They simply shift the bits from 1.2V core voltage to 3.3. They do not effect the analog reference voltage. In fact the reason it is imperative that the AVCC be an extremely low impedance is because the frequencies involved are extremely high. All the VDD supply has to do is maintain enough voltage to keep gates of the Qs saturated. The AVCC supply is crucial because it has to absorb and source current at very high frequency and with very low noise at the same time.
Now it is important to bypass the VDD pins well (as I have done) but these pins are not in any way tied to the analog reference voltage.
The key to good results is a clean very low impedance AVCC supply.
The 1.2V regulator
The 1.2v supply powers the core of the DAC and it is therefore a pretty important supply. The recommended regulator is the widely used and low noise ADP151-1.2 and can supply 200 mA maximum.
The operating current at 100 MHz clock frequency (160 mA) is already very close to the max that this regulator can provide. In addition, if higher sample rate material is used, then we are getting very near (or at) the max operating current.
I think a more capable regulator is a good idea. I like the LT1963A which can provide up to 1500 mA. (we audio diy types like overkill ). The adjustable version can be configured to provide to 1.21V and it is also the lowest noise configuration for the family.
Noise is 14 uV RMS (compared to the ADP151 at 9 uV RMS). Page 18 of the datasheet [link] says:
The LT1963A regulators have been designed to provide low output voltage noise over the 10Hz to 100kHz band-width while operating at full load. Output voltage noise is typically 40nV/√Hz over this frequency bandwidth for the LT1963A (adjustable version). For higher output voltages (generated by using a resistor divider), the output voltage noise will be gained up accordingly. This results in RMS noise over the 10Hz to 100kHz bandwidth of 14µVRMS for the LT1963A increasing to 38µVRMS for the LT1963A-3.3.
Compared to the LT1763 (used in Buffalo II) it is actually lower noise at the lower end of the frequency scale (<1KHz) and almost the same at the higher frequencies. At, say 100 Hz, the noise density of the LT1963A is 40 nV/SqrHz and for the LT1763 it is near 300 nV/SqrHz.
The LT1963a is also designed for Fast Transient Response. I’ve looked at many datasheets, the LT1963 seems the lowest noise of the “fast transient” LDOs. Fast transient is specifically designed for powering digital cores such as DSP, FPGA and in this case the digital core of the ES9018.
Modding the board and regulator
The challenge here is to install it in the board since the footprint is for the ADP151.
Comparing the pin assignment of the ADP151 and LT1963 we realize that they are kind of mirror image of each other.
For 1.21V operation, the ADJ pin in the LT1963 is tied to the OUT pin. SHDN is the EN(able) pin.
Flipping the LT1963 and bending the pins the other way should do the trick.
Scrape the solder mask to make the pads.
Checked the ground pads to determine the correct pin orientation. For 1.2v operation pins 1 and 2 are shorted. Pins 3 and 4 can also be shorted. Made pads for pins 5 (input) and 7 (GND). Pin 6 and 8 can be connected with wires.
The wire connects the enable pin to high (input voltage)
These are the bypass capacitors I used. The regulator output bypass is 100 uF with 30 mOhm ESR. According to the datasheet a minimum of 5 mOhm for a 100 uF capacitor is required for stability and reduced ringing. There is an ELNA 1000uF on the input side.
Works as specified: 1.21V. I used two alkaline batteries as input voltage and 10 ohm resistor as load (120 mA of current)
IMPLEMENTING THE 3.3V AVCC
The conventional wisdom is to use a shunt regulator for low noise and low impedance.
I think I will use the ultra low noise TSP7A47 series regulator from TI and add capacitors in parallel to the bypass capacitors for the AVCC supply lines to improve the transient response.
AudioLab uses such a configuration:
Here is a mockup with 5 mm capacitors:
Here is a good reference post from diyaudio [link]
We’re talking about basic high speed digital design, i.e. we want the logic 0/1 to arrive at the destination (where it matters) at the right levels and at the right time.
The uni-directional (transmitter –> receiver) signal is as simple as things can get and is what I2S runs on. You worry about reflection when the wire/trace is long and/or when the signal risetime is fast (e.g. a simple reset signal from a FPGA can have issues). To help alleviate that, series termination (located at the transmittter) is the simplest form. Your PCB stackup and trace width/separation should also be design to match the impedance characteristics (e.g. 50ohms usually for single-ended, USB is 85ohms differential).
Now how do you know if your circuit is well taken care of? You look at the signals with a scope (proper probing required). You want to ensure that the signal rise/fall edges are monotonic, under/overshoot is within spec and crosstalk from adjacent signals are acceptable. The first 2 parameters are achieved by proper high speed layout techniques (use termination, proper pcb traces, minimal via transitions, no routing over plane splits, etc). The last parameter is achieved by proper pcb layer stackup design and wise routing.
Do NOT EVER put caps on digital transmission lines. Series caps are typically for AC coupling (more commonly seen with PCIe than I2S). Parallel caps can snub terminations but the danger here is that they slow down the risetime of the signal and can lead to loss of timing margin (data setup/hold).
Series termination are used successfully for far more complex stuff, e.g. SPI at 50MHz, DDR2/3 at 500MHz. Typical values are 22R-33R. I have never seen 47R series termination resistors in any embedded design. I was just at Embedded Systems Conference West in San Jose. Saw plenty of reference designs, including I2S/digital audio stuff. Nope, no 47R there.
Look up Dr Howard Johnson’s books (the digital designer, not the hotel chain). Lots of good info.
For the typical hobbyist like the OP with a simple I2S circuit, I’d just put my chips as close together on the PCB as possible and call it a day. If you like to cable I2S from one board to another, then that is where you’ll run into issues. Most important point for cabling I2S is to ensure adequate ground returns for every signal (e.g. you use ribbon cable, put a GND wire next to every signal, use a 2-row connector with one row being all GND pins). Consider active buffering if your cabling is long.
(Update II 01/08/12: Using Hifiduino s/w and Arduino controller)
The current version of the Hifiduino s/w [link] supports the following configurations for the board:
1- The board in its default configuration which is I2S and SPDIF in Data 7 and Data 8 simultaneously
2- Inputs modified to match the input wiring for Buffalo II: will support both I2S and DSD with an input board such as Amanero and also SDPIF on Data 1 if not using I2S
3- Inputs modified to support “smart” wiring: will support both I2S and DSD with an input board such as Amanero and also SDPIF on Data 1, Data 7 and Data 8 (SPDIF in TTL levels – 3.3v)
- SPDIF inputs are TTL level. You cannot just take the spdif output from a consumer device such as a DVD player.
- Unused inputs should be grounded
- All inputs should have a termination resistor
(Update 01/08/13: More on using the DAC in voltage mode)
Using the DAC in voltage mode means not needing to populate the opamps. According to the designer of the DAC:
… The current mode is simply when the current going in and out the pin of the chip is being sensed. This mode has the benefit of cancelling 2nd and 3rd harmonics of some of the internal analog circuitry.
The “voltage mode” is when the pin of the chip has a voltage that is being sensed. While this has the 2nd and 3rd harmonics (at the -100dB level or so), some people have even claimed this mode is more “tube-like”. It is all personal preference. [link]
(Update II 01/06/13: Buffered differential output)
The board only implements output pads only for single ended output, but since this is diy, you can take the output basically anywhere you wish.
The buffered differential outputs can be taken off the outputs of the opamps. By implementing the opamp I/V converter, you “force” the DAC into current mode allowing for better THD.
And then add the low pass filter specified in the ESS documentation
(Update 01/6/13: Correction on reset circuitry)
Seller sent me a message indicating that the reset circuit must be populated or else the DAC will be unstable. This makes sense as the pin is floating if none of the components are installed. I’ve corrected the instructions below.
(Update II 1/5/13: Differential Vout, chip address, AVCC/2 offset, I/V)
I am a fan of using the Sabre32 DAC in differential voltage out mode. I have been using the Buffalo II DAC this way since day one and still do not have the motivation to add an I/V stage . There has been comparison of this DAC in voltage mode against other DACs and I’ve documented some here: [link].
BTW, if you are concerned that only an optimally routed fabricated and designed board can produce good sound, you should take a look at what Mr Abraxalito is up to: [link]
I plan on using this DAC in voltage mode also, at least from the beginning. You can take the raw differential outputs as shown below and take the GND from any nearby GND pads
The chip address is already set at “o” -connected to GND. If you are using the board in stereo mode, there is no need to do anything here. For mono operation you would have to cut the trace for one of the boards and connect the address pin to VDD (set to “1″)
The 2 10K resistors establish the AVCC/2 offset voltage that is used by the I/V stage
The I/V output stage is an implementation of the circuit in the 2-channel eval board [link]
The schematic calls for 1 uF bypass for the power supply lines and voltage offset lines. Probably a good idea to use a 0.1 uF SMD bypass together with a 10 uF electrolytic bypass.
(Update 1/5/13: Reset, power supplies)
Two external supplies are to be provided: a supply for the analog section and a supply for the digital section. The 3rd power supply is the 1.2V and it is provided by a local regulator that must be installed (I believe the current boards already come with this regulator installed).
The 3.3V analog or AVCC supply is the most critical. Preferably, choose a low noise shunt-type. I have yet to decide what to use. I may built up some of the Placid V1 boards I have.
For the 3.3V digital, I plan to use the TPS7A47 Eval Board.
(Update 1/4/13: Chip bypass)
I’ve traced the power supply bypass positions and labeled them. There are 8 bypass locations for the most important supply the 3.3V analog (AVCC), 3 bypass locations for the 1.2V and 2 bypass locations for the 3.3V digital supply. I am not sure if the 1.2V locations would just require a 10 uF bypass (which are indicated in the ESS available documents) I think having a 0.1 uF SMD together with a 10 uF electrolytic on the other side is a better bypass.
Every AVCC pin has its bypass capacitor, but not every 3.3V Digital and 1.2V has its own bypass capacitor. Doesn’t seem to be enough space (or required?), but the seller did put the bypass where it most counts, the analog AVCC
(Update 1/4/13: more photos)
Front side of entire board. Since I probably got the first V2 board, the ADP 1.2 regulator was not soldered… The solder pads are all silver in color. The coloring is reflection of the surrounding, including the sky…
Back side of board
Input connections details (front and back)
Notice that D5, D4, D3 and D2 are tied together for default stereo I2S operation and connected to the front “data” pad. In order to use the inputs separately, one needs to cut them.
Just received the board. Very good quality, some photos below… This is the V2 version of the board. This version came quickly after V1 and (to my surprise and delight) implemented many of the things I suggested in the diyaudio thread. That is pretty good response from the creator of this board. In this eBay website, he mentions that this design will keep on evolving, thus I would encourage anyone interested in this board or future iterations to provide feedback and suggestions at his diyaudio thread.
It is good to see another DIY version of this DAC especially at this low price. But this requires the most work. Keeping in mind that the chip alone would cost $60, the price of this DAC board is near cost. So if you are handy with soldering small components, this board should provide plenty of fun and audio satisfaction.
All the input lines are available (e.g.: not grounded) -this is an improvement from V1. The output is stereo, so it can be used as stereo or mono but not 8-channle. However, having all the inputs available, we can leverage the internal mux and auto-detection to switch between different inputs.
There are pads for U.FL connector for clock input on the other side of the board. This is also an improvement from V1
Pads seems fairly easy to solder. I believe it can accommodate a 1206 SMD capacitor or a 5 mm-leads radial capacitor. Footprint for any size oscillator including the Crystek CCHD-950/957
This gives you an idea on the dimensions of the solder pads:
The first thing to figure out are the power bypass capacitors. The board has space for both SMD capacitors and through hole capacitors. One idea is to mount both of them (of different values) in order to improve filtering:
According to this application note from TI [link]
The most common values bypass capacitors are: 47 µF, 22 µF, 4.7 µF, 0.1 µF, and 0.001 µF. The higher value capacitors (47 µF and 4.7 µF) work well at relatively low frequency (low-frequency bypass). The 0.1 µF targets the middle frequency range, while the 0.001 µF or smaller capacitors handle higher frequencies (high frequency bypass). Choosing two or three capacitors with different capacitance ranges will effectively filter a wider noise bandwidth.
Seems using a 0.1 uF SMD capacitor with a radial 0.001 capacitor is an excellent idea…
I hope there are tube fans reading the blog. If not, do not worry. This is the only tube device I have . You can read part 1 here: [link]
In reality, I am not really that much of a fan of tubes but this amp has served me well in my secondary system and consistent with the theme here (“lot of value, little money”) the ASL Wave-8 is great value in audio (unfortunately no longer manufactured). But the mods are a lot of fun to do and improves this already good amp up “a notch or two”.
Even compared with the current, higher power version of this amp (the Wave-25), the Wave-8 still has “sweeter mids” according to the designer. The bigger brother of course has more power and extends further in its frequency response, but with mildly efficient speakers (the KEF Q15.2 I use are 91 dB), this is the amp to have.
Two primary mods for the ASL Wave-8 were described in the previous post, along with rearranging the ground connections:
- Better regulation of the power supply
- Better coupling capacitors
The PS mod is the most elaborate. Based on simulation, the ripple is reduced from 2V to apporx 20 mV, an improvement of 100X. A we know, the quality of the power supply is perhaps the most important factor for good sound.
Based on simulations (Duncan’s PS Designer II -Scott sent me the simulation files) we can compare the output of the two power supplies:
Original PS: ripple ~ 2.4 volts
Modded PS: ripple ~ 27 millivolts
Hum has been a problem with this Amp since day one. Many people have reported hearing hum and mine are not an exception. However it is low level and only audible if you put your ear close to the speaker. After the mods, which included rearranging the ground connection, I can report that the hum is much, much reduced. With a RS sound meter and only after putting the meter inside the port, I can measure 57 dB on the modded amp and 66 dB on the original, un-modded amp. If I put the meter right next to speaker surface (or anywhere near), the meter (which has a sensitivity of 50 dB) does not measure anything on the modded amp.
This is a reduction of 9 dB which translate to about 80% reduction in sound power. Very nice. According to some, it may not be possible to totally remove the hum in this design:
My experience has been that getting all the hum out of a push-pull amp can be difficult. When one tube of the push-pull pair is drawing more current than the other tube there will be a hum. I can hear this hum from about 1 foot from the speaker. I usually set the DC balance to try to minimize it. The Wave8 does not have a DC balance, so all you can do is to swap output tubes around such that each tube of the push-pull pair is producing equal amounts power. [link]
But more can be done…
There is an excellent tutorial on heater wiring. Essentially: heater wiring – the Good the Bad and the Ugly
- Tight twisting of the wires -to cancel the magnetic field (shielding reduces electric field but no magnetic field)
- Move wires away from circuitry
I redid both the AC wiring and the heater wiring. The AC wire goes from the back of the amp to the front where the switch is located. Originally I used stranded zip-wire with only 6 turns along the length of the wire. This time I utilized18 gauge solid core and twisted it very tight (at least 6 turns per inch). Solid core is better because it keeps its shape. [Tip: you can buy thermostat wire which has two 18- gauge wires for 24 cents/foot at the Home Depot]. Same for the heater wire. The first photo is the mains AC switch, the second photo is the filament/heater wire (which is also AC)
I measure the sound level as before and now I read about 56 dB. We have an improvement of about 1 dB. The original wiring was pretty good. In any case, a reduction of 90 % of hum (10 db) is already pretty good.
HEATER WIRING: we can do more
Here is the original heater wiring:
Here is a diagram and instructions from http://www.el34world.com/charts/commonhookups.htm:
If you do not have a heater center tap on your power transformer, you must run two 100 ohm 1/2 watt resistors to ground to create an artificial center tap. If you do not have a center tap, you will get 120 cycle hum. Each 100 ohm resistor is soldered to one of the heater wires. The other ends of the 100 ohm resistors are twisted together and then soldered to ground.
The heater wires are usually run up in the air, above the tube sockets in a twisted pair. Twisting the heater wires cancels hum. This is why phone line wires are run in twisted pairs. The twisted pair wires drop down and get soldered to the tube socket pins. The twisted pair continues down the line to every tube in the chain.
Keeping the wires in phase helps with hum sometimes. In other words, pin 7 on one power tube goes to pin 7 on the next power tube. Pin 9 on a pre amp tube goes to pin 9 on the next pre amp tube. EL84 power tubes heater connections are pins 4 and 5. Most other 8 pin power tubes use pins 2 and 7.
Also, there are some discussions on heater wiring with PCB (as opposed to point to point)
There is NO good way to get low hum with PCB heater wiring.
Live with it, or fix it. First verify that 6VAC flows in PCB “wires”, simple circuit tracing. If so, cut the heater lines AT the socket pins, leaving only the pin solder-blob. Get some hookup wire and run heater power OFF the PCB, in twisted pair cable AWAY from all audio points (basically everything except the heaters). Study some of the excellently-wired amps posted here. [link]
The original heater wires are connected “out of phase” (don’t know why. Perhaps to keep the wires as short as possible). Also, the artificial center tap is done at the end of the wire run with the ground connection at the end of the ground line. This means the current travels the whole length of the ground trace to the star ground. Will try implementing the artificial center tap near the power ground and also connect the wires in-phase.
Notice that the artificial center tap in the PC board has been removed (R18 and R19) and the heater wire has been shortened, more twists and connected “in-phase”:
The artificial center tap is now implemented towards the back of the amp near the transformer, with direct ground wire to chassis:
Results: I measure 55 dB, an improvement of 1 dB.
ELEVEN dB REDUCTION IN HUM
All the AC and heater wiring modification resulted in about 2 dB reduction of hum. Not too bad. The total reduction combined with the previous mods is in the order of 11 dB. Remember that this measurement is sticking the meter into the port of the speaker. If we measure the sound pressure at the speaker, we measure 57 dB with the un-modded amp and unmeasurable for the modded amp (which should be 46 dB, 11 db lower).
How loud is that? I can start hearing the hum at about arm length (~32 inches) from the speaker with the unmodded amp and at about 7 inches with the speaker with the modded amp. This is with good ears and in the quietness of midnight… In the morning hours, the distances are about 24 inches and 3 inches respectively.
Yeap, it is a tube amp. The Amps are push-pull mono design and are approx 11 years old. They were introduced as “best bang for the buck” for $99. They became an instant hit especially with the diy/modder crowd. Unfortunately, many off the documented mods have disappeared from the web. That was before the time of free blogging . The last known troubleshoot/mod report is from 2009 [link]
They also got great reviews such as this:
At an entry-level price of only $99 per side, few power amps compare to the amazing rendition of high-end audio that these charmers give. If there is an entry level tube amp out there for neophyte audiophiles, or simply some one who wants the most “tube bang for their audio buck” with their above average efficiency speakers, these have got to be the ticket. They certainly punched mine. [link]
Let me begin by talking about the amp these replaced. I previously had a stock Phase Linear 400 II. 200watts/ch. I kept the Phase Linear in my rack to compare but I liked the Waves so much I didn’t even bother to check the Phase Linear for a month. For me, audiophile nervosa is a powerful force and the fact that I didn’t do a “reality check” back with my old amp for a month speaks volumes about the difference. [link]
Straight from the box these amplifiers (given of course proper burn in) will impress you. They will never impress those with thousands of dollars invested because hey how could something that cheap be any good? Just keep it your little secrete so all the other folks who love sound but not high prices can enjoy them. We don’t want Antique Sound Lab to start selling them for what they are really worth now do we? [link]
At this price the company was probably just breaking-even. And anything that had little to do with the sound was spared. For example, the power and output transformers covers are just raw steel without any kind of finish. But even so, if you were to buy these transformers in the retail market, they would probably set you back $40 a piece, so likely they manufactured their own transformers and probably everything else for the absolutely minimum cost.
I have been using them in their original factory-stock configuration for my bedroom system connected to a Sony SACD player through a passive volume control and powering KEF bookshelf speakers. No complains so far but having been presented with the opportunity to get expert advise in modding this amp, I couldn’t pass the chance. That expert advice came from Scott [Scott17 at diyaudio].
After a few conversations with Scott, I realized that he was selling tube kits. So I asked him for a few modding tips for my amp. Scott was extremely generous with his time and knowledge and after a few exchanges where I described the innards and measurements of the amps, he developed a complete plan for me. In a way he was even more excited about this project than I was, often expressing his excitement and anticipation for the results. I can only imagine the level of support and care he gives to the people who buy his kits.
I have also scanned the full factory documentation and posted it here: [Wave8 Manual]
Here are the innards of the Wave-8 fully stock from the factory:
Fairly good components: film resistors, name brand capacitors, spacious PCB, ready for modding. The tall blue power supply capacitors are high voltage Nichicon VX electrolytic capacitors. I think those models have been discontinued as the current configuration for VX capacitors are axial rather than radial. According to this document [link], the VX have been replaced by the VR capacitors. The ones used are Nichicon 450V 47 uF and are good standard capacitors at $2.58 each [link] (the equivalent from the new series).
The high voltage DC is provided by the 4 diodes and a CRC filter of C9, R17 and C8. C9=C8=47 uF, R17=100 ohm
MODDING THE POWER SUPPLY
Scott suggested that the most noticeable improvements would come from modifying the power supply section. The following mods were recommended (in order of importance):
- Replace R17 with 4-Henry DC choke: [link], making a CLC filter which provides better filtering than a CRC filter
- Replace C9 with 630V 47 uF film capacitor (used 400V 40uF instead): [link]
- Replace C8 with 450V 220 uF Nichicon KX electrolytic capacitor (used 400V 180 uF KX instead): [link]
- Replace the diodes with ultrafast recovery types (UF4007) [link], further reducing noise
Why use a choke? Why not just a big series resistor?
A choke is used in place of a series resistor because the choke allows better filtering (less residual AC ripple on the supply, which means less hum in the output of the amp) and less voltage drop. An “ideal” inductor would have zero DC resistance. If you just used a larger resistor, you would quickly come to a point where the voltage drop would be too large, and, in addition, the supply “sag” would be too great, because the current difference between full power output and idle can be large, especially in a class AB amplifier [link].
The first thing you realize, is that film capacitors are “gigantic” in size. DC chockes are also large in size and even the higher value electrolytic capacitors would be difficult to fit. So before you embark in buying the components, it would be wise to figure out where to put the components and whether they will fit inside the case. After doing some mockups and measurement, and based on stock availability from the different supplies, I ended up with the components specified above. The clearance from the bottom of the PC board to the edge of the chassis is 40 mm. So most components would have to be 40 mm or less in one of their dimensions.
I did several mockups to see if the components would fit. here is an earlier mockup. Eventually ended with a single 40 uF film capacitor instead of two 20-uF capacitors by doing some “clean-up” on the PC board.
“CLEARING THE PCB”
I notice that there is a lot of space in the PCB board and that the layout can be made more efficient. The 40 uF film cap I selected would fit well in the back of the board by moving some components. The first thing I did was to move the fuse, switch and associated wiring to a small board and the resistors I re-installed in the backside of the board.
Actually, it is way better to have the input AC wiring off to another board. You move a lot of wires out of the way.
Although not recommended in the mods, I replaced the IEC socket with one with built-in EMI filter.
C8 and C9 would also be replaced, so we can remove those too. Now we have a nice empty space where we can install the large film capacitor.
The diameter of the film cap is 40 mm. Any larger would not fit in this location.
Replacing the diodes with ultra fast recovery diodes and installing C8. C8 is installed in position C9 (to free up space for the large film cap, which is C9). This requires that we cut the + trace and find a place to connect the + terminal of C8.
The KX series are designed for audio use.
C8 is in position C9. We need to cut the trace as shown and connect to the + connection of C8 (short orange jumper – you can see the trace goes around and connects to + of C8 position)). The long orange jumper connects the + connection of C9 to the large film capacitor which replaces C9 (not in the picture).
Installing C9. The 47 uF electrolytic capacitor is replaced with a film capacitor. Because of the size, I used a smaller-sized 40 uF 400V film capacitor. The + connection of C9 connects the + connection of the 4-diode bridge rectifier.
According to this article [link]:
One of the best amplifier power supply grounding schemes is a “star” ground system, where all the local grounds for each stage are connected together, and a wire is run from that point to a single ground point on the chassis, back at the power supply ground. Even better is a two-point star, where the power supply grounds (PT center tap, first filter cap ground) and output stage grounds (output tube cathodes for fixed bias, or cathode resistors for cathode biased, and output transformer secondary ground) are connected together and to the chassis at a single point, right at the ground of the first filter capacitor. The ground of the second filter capacitor, after the choke or filter resistor, is the star ground point for the preamp stage grounds. Use a local common point for each preamp stage ground, and run a wire from this common point back to the second star point. If two stages are out of phase with each other, the can share a common local ground, but don’t use more than two stages per local common ground. This concept can even be taken further, with multiple star points for various amplifier stages.
I used a single star ground approach (since the ground of the second filter capacitor is right next to ground of the first filter capacitor) with the negative lead of the film capacitor as the star ground. To this ground connects the negative output of the 4-diode bridge rectifier, the speaker GND connection and the GND from the “front-end” where the input signal ground is connected. From here, there is a single cable that connects to the chassis.
It is important that you pay close attention to grounding. This means good solder contacts, good contact to chassis (scrape the paint off) and good wiring. Make sure you measure continuity and resistance from all the ground points.
The choke is installed to the side of the chassis. The choke replaces the 100 ohm power resistor.
“This is the most common 90mA DC rated Fender* type choke used in many of their amplifier applications. Provided with shields for extra protection. Paper layer wound choke like vintage era originals!”
A bit of “Made in USA” in a Chinese Amp
The choke, the film cap and the electrolytic capacitors are near flush with the edge of the chassis.
REPLACING THE COUPLING CAPACITORS
After the PS mod, the coupling capacitors will also make a big difference.
Scott recommended the use of ERSE caps as an economical upgrade of the coupling capacitors.
The ERSE replaces the BENNIC capacitors.
Change R14 and R15 with lower noise versions. The cathode bypass caps C6 & C7 should be changed to 220uF. The stock configuration of 390R/100uF yields a -3dB corner frequency of 4Hz. As a general rule, the -3dB corner frequency should be 1/10th of the desired low-end response of 20Hz, which is 2Hz. This mod will achieve just that.
THE COMPLETED BOARD
Before and After
More bypass capacitor options:
There are so many choices and reviews. According to reviewers, their performance also depends on their application.
Capacitor review links
According to Joseph Lau, the designer of the amp, the calculated value of the bypass capacitor is 0.1 uF. 0.22 uF is already overdesiged. [link]
|Bennic XPP 400V||~18×10||<$1|
|Erse MPX 630V||20×11||$1.37|
|AmpOhm tin foil 630V||54×25||$20.55|
|AudioCap Tin Foil PPT Theta||30×17||$8.06|
|ClarityCap SA 630V||21x1928x20(.47 uf)||$3.70|
|ClarityCap ESA 630V||20×24||$7.90|
|Multicap PPFX 400V||28×15||$6.45|
|Russian Teflon FT-3 600V||72×31||$5.50|
|Jantzen Z-Superior 1200V||43×22||$8.64|
|Audyn Plus 1200V||43×25||$6.50|
Even more mods [link]
Don’t use the cage as it makes things sound worse however the IEC allows you to use a quality after market power core and that is well worth the few extra dollars.
The parts count inside is very good and I did not feel that replacing caps and resistors would yield a big enough difference to justified the expense. The amp can and does respond to bypass caps and these are most cost effective and easy to install. Bypass the power supply caps with 1000 Volt ceramic disk caps.
All large value caps were bypassed with 0.1 uf plastic caps (despite what people say Mylar sound fantastic). All the small value caps were bypassed with 0.01 uf plastic caps.
I removed the “bell” or “end” caps from the output and power transformers. I then installed an electrostatic copper wrap shield around both of the transformers and in my case turned the output transformer 90 degrees to the power transformer and re mounted both transformers on rubber grommets to reduce vibration to the chassis. Removal of the “Bell” caps really opened up the sound and gave a sense of life and new dynamics. I have found this to be true on other amplifiers also. I did not do a direct before and after comparison with the physical orientation of the two transformers however with two chokes you want them aligned this way to place one in the exact electrical/magnetic “null” of the other an so minimize coupling between the two. This makes sense so I decided to do it with the two transformers even though it did involved making some of the lead wires longer.
The second last change that I made was to replace the input interconnect and to shield the power on/off wires which run to and from the front of the chassis to the back to the chassis. This one is up to you to do as you see fit.
Last but by no means least I froze the tubes down to liquid nitrogen temperature. If you haven’t tried this do so as it is a great upgrade all on it’s own. Sound is smoother and there is more detail with greater resolution and an apparent increase in dynamics. I can assure you that once bit (frozen) you will never go back to non frozen tubes or other parts for that matter.
Next go to Elliott Sound Products and read his article on a passive volume control “project 01″ build one to use with your “Wave” amplifiers and then you will enjoy a level of performance you probably have never dreamed you even reach. The price of entry is a laugh. You will now be able to make your “Audiophile” friends so sick with what they own and with what they paid for it that they will in the words of J C Morrison “want to go and find a high place from which to throw off their gear”.
And more mods [link]
(These recommended by Joseph Lau, the designer of the amp)
I then started thinking about resistors. Joseph Lau recommended replacing 4 resistors in the signal path per unit. R4, R7, R10 and R11. Those are a 33k, 390k, and 2 1ks. I thought, what the hell and I ordered full compliments of Rikens, AN Tants, Holcos and Roederstein Resistas from Angela.com, Michael Percy, and Audio Note North America.
I began by replacing one amp with all Rikens. The bass response increased dramatically and I liked that (Tom Waits never sounded more gravel-ly and unshaven), but after closer examination I felt the highs were rolled off and there was way too much of a “tubey” or “hollow” sonic characteristic. Also some low end distortion.
I switched out the Rikens for all Holcos, but that didn’t seem exactly right either. It swung the other direction and I found it was too lean for my taste and lacked deep dynamic punch. Tom Petty whined too much.
I then put in all AN Tants which were very musically satisfying in the bass but the top end was restrained, or oddly veiled, but better overall than the other pure attempts. Ry Cooder had post nasal drip. hahaha
I then started thinking that I should try combinations, looking for the optimal blend. I put in the 33k and 390k Rikens for bass, with 1k Holcos in R10 and R11 and wow. Very nice highs, not too bright, but “there” with deep bass. I decided to try a variation on the other amp which was AN Tants in the 33k/390k positions and Holcos in the 1k positions and then duel the two off, A-B.
Upon extended HEAVY DUTY listening, there was more bass with the Rikens, but the Rikens were muddying or smearing the bass somehow, while the Tants were producing very slightly less bass, but it was a VERY well defined and musical bass. There was no comparision. The more I listened, the more I got goosebumps, and the more I loved this combination. AN Tants in the 33/390 positions and Holcos in the 1k positions created tonal characteristics nothing short of spectacular IMHO.
All the music I tried sounded “just right” in tight bass and extended yet not overly bright highs. Sampled music included Tom Petty Wildflowers, Ry Cooder Bop ‘Till You Drop, Les Nubians Princesses Nubiennes, Sheila Chandra Roots and Wings, Tom Waits’ Foreign Affairs, Eric Truffaz The Mask.
I did pop in 1k Roederstein Resistas in the 1k positions in one amp to test that possibility that they would increase dynamics, but they were a bit dry and substantially less dynamic than with the Holcos in the 1k slots.
There certainly are other combinations I didn’t try, but I think I’ve found a real winning combination that bring these little Wave-citos up a few big notches that have them competing sonically with my Zen Select amp for tonal character, but they’ve got muscle behind them.
I did find that when I put in Caddock TF020Rs in my volume shunt on my Foreplay Preamp, it made a huge difference and was the best I had tried (over Rikens, AN Tants, Holcos and Resistas). I may go ahead and get some Caddocks for the 1k slots, but I’m pretty certain that the AN Tants are the bomb in the 33/390k positions as they seem to tame the other stock metal film resistors which tend to be so bright, while the 1k slots keep the sound open and let in the high dynamics.
I’m no expert and I’m using “down home” language to try to explain my experiments and what I’m hearing. My Waves are singing beautifully.
I’d love to hear what anyone else has done to these fun little amps.
Overall, a very fun project. At about $50 per amp (and free guidance from Scott), it is a heck of a good deal…
Part II of the mods here: [link]
THE SRA2.1 (“SHUNT REGULATOR AVCC 2.1″?)
Superb workmanship and finish.
Here is the “SRA1″ version that was bundled with the Buffalo II DAC (the one I have)
Here is the “SRA2″ version that was available prior to the introduction of the AVCC 2.1. This “SRA2″ version has been available for quite a while, at least since the the BII switched from an 80 MHz clock to a 100 MHz clock. Notice the thicker traces and different circuit topology (gone are the n-channel transistors QN1 and QN2)
Here is on a BIII
Here is the “SRA2″ on a BII-100 [link]
Boy, my AVCC is two generations old!, and I didn’t even know it. Maybe because it has been hiding on the bottom of the DAC board
Lets start with the “ideal opamp” characteristics:
- Infinite voltage gain
- Infinite input impedance
- Zero output impedance
- Infinite bandwidth
- Zero input offset voltage (i.e., exactly zero out if zero in).
|Parameter||LMP7732 (Old)||OPA2209 (New)
||Data sheet has graphs but they use widely different units, so it is hard to compare.
|Input offset (max)
||0.006 mV||0.035 mV
||Seems the old one has an edge for these first 5 parameters
|Input Voltage Noise Density
||Here the new opamp has an edge. This seems a critical component for good performance [link]
||Here the new opamp has a slight margin for stability
The comparison above does not give a clear indication as to why the new opamp is superior to the old one. I am sure the enhanced performance is in the circuit design and choice of component values. Without doing a circuit simulation, and testing/measuring, is is not possible to say which one is better. We trust, however, that TPA has made the right selection for opamp.
Indeed, as Russ has commented,
Bottom line is that the old op-amp was excellent on paper (and probably for other applications), but was not so good for AVCC in practice. It was also strangely finicky, meaning it could work perfectly usually – but occasionally be upset just by changing rail voltage or applying the right kind of external stimuli or load.
The output voltage of the AVCC starts at 3.6V. After the LEDs warm up, the output voltage settles at around 3.56V. This is typical of regulators that use LEDs as references. Input voltage is 6.2 V. The load resistors for the test are 72 ohm. The output current is therefore 3.56/72= 50 mA.
At first I had use a 33 ohm resistor. This load was pulling 108 mA and the voltage dropped down to about 2.5V. The regulator cannot source more than 100 mA (as specified).
Compared with the V1 (shown below), the LEDs are evenly lit.
HURRY AND UPGRADE
Original Buffalo IIs (the ones which came with the SRA1 AVCC supply) should benefit the most from the new AVCC 2.1 because both the circuit configuration, layout and opamp has been updated/upgraded. NICE!
If upgrading from the later 2.0 version, the layout, opamp has been updated/upgraded and the circuit has been tweaked with the compensation capacitors. Nice upgraded but not as nice as upgrading from the original BII
I performed a frequency response plot for my modded D3V2. The red response line is a Behringer USB DAC I used to obtain a baseline response curve. As you can see it is more or less flat from 20-20KHz. The blue response line is the D3. Yes it does roll off a little, about 3dB, starting at 10KHz. I doubt that is audible. Certainly not many loudspeakers are any better. Not bad for 20 bucks or so.
Gauging form the number of comments on the “Inside FiiO D3” post, it seems there is a lot of interest in this low cost device.
Mr Scott sent me a few photos of his modded D3V2. The FiiO D3V2 is the infamous version that many complained of having rolled-off highs. But according to Mr Scott,
My experience with the D3 V2 is that I did not have any complaints with the stock unit that I received. After the mods it has definitely improved with regards to clarity and soundstage. Even though I have 56 year old ears, I’ve been a musician for about 40 years, I design and build top end hi-fi tube amps, and have been a discerning listener for a long time.
Thus, contrary to others experiences, this device seems to operate properly. In any case, here are the photos of his modded D3
Also replaced the zener 3.3V regulator with an ST Microelectronics LE33CZ 3.3V regulator.
If anyone else wants to do this, you must remove the zener as well as the two 75 ohm paralleled resistors that are connected to the cathode of the zener. Then you connect the ground of the LE33CZ to the pad where the anode of the zener was, the “in” of the regulator goes to where the two 75 ohm resistors were, nearest the edge of the circuit board. The “out” of the LE33CZ goes to the pad where the cathode of the zener was.
Here is a closeup of the 3.3 regulator
After listening to this with the mods, I can say that the stereo separation and “spaciousness” (for lack of a better word), has definitely improved.
BTW, Mr Scott is “Scott17″ at diyaudio [link]