...the end of seperate DAC + AMP decade might be near!!! ;)
For those of you who got tired to run after every new DAC and amp (I did), and
those who've read the 1000th review of another miracle DAC (I did), or those
of you who got more confused then enlightened with the endless number
of DACs of choice out there (I did again), or those who look for a very small,
cost efficient and still great sounding system ... (Oh -- I'm talking about myself..)
Appendix 1: Market overview
Appendix 2: Patent
Appendix 3: DDX320 Modifications
...I'll recommend to look for an alternative solution instead: (I did. ;) )
Full Digital Amplifier (FDA)
What is a Full Digital Amplifer?
Full Digital Amps run direct PCM (Pulse Code Modulation) to PWM (Pulse Width Modulation) conversion.
The company called STM holds a patent on the PCM/PWM amp since 2003. You'll find interesting information in the patent about how such an amps works.
There is no low-level lossy analog conversion prior to amplification. There are even models which run DSD (SACD) to PWM.
Sometimes Full-Digital amps are called Power-Dacs. That can be
misleading. typical DAC converts PCM digital to analog.
The FDA digital to digital!!!
You'll find products called Power-DAC/Digital Amp, where
the manufacturer integrates a DAC and a seperate amp in one box.
That's not what we're talking about here. We talk PCM/PWM conversion
done in one chip.
The "unfiltered" output of a FDA amplifier would be digital pulses ( as in class-D) with a changing pulse width.
The frequency of those pulses is pretty high (a couple of hundred kiloherz) above the audio band.
Just a simple passive 2nd order lowpass filter is required to filter out the modulation frequency and to recover the audio signal properly.
Usually another filter (Zobel filter) is added to filter out high frequency artefacts.
The FDA and Class-D amp output stages are identical. There are certain small issues related to this filter. The most evident flaw is that the output filter interacts with the speaker (same as with pretty much all Class-D amps).
Impedance shifts throughout the audio band of the speaker or a poorly matched filter (speaker/amp) slightly influences the resulting signal quality.
Best results can be expected with impedance linear speakers and matched amp output filter- speaker impedance.
Though the audible differences running e.g. an output filter dimensioned for a 4R speaker on a 8R speaker will be close to neglectable. That's my experience at least.
The advantages which come with those amps (read on) make those "minor" issues look irrelevant.
Newer FDA models let you adjust the output filter impedance to match your speakers.
However. People report minor changes only, when trying wrong filter settings. ( matches my experience)
Some FDA models come with low-power (20W) one-chip solution, where the power-stage is inside a single chip and some others come with a 2 two-chip solution where the external PWM power-stage accepts a PWM signal from the driver chip on the input.
What you'd go for will depend on your speaker efficiency. On a >94db/W SPL speaker, the 20W amp will do a pretty good job.
But even on my 98db/W SPL speakers I prefer the >50W.
Let's put up some advantages of such an amp technology .
What you'll gain running a FDA:
NO MORE bothering about, respectively elimination of, error sources such as:
* DA conversion at a pretty early stage, thus reduction of analog signal losses.
(Digital transmission is lossless from an audio signal perspective under normal conditions.
Analog transmission is never lossless.)
* DAC output-stages (active, passive, transformers, tubes, opamps, asf. - read the endless
discussions about it - I did)
* AMP input stages (another opamp with feedback, another capacitor in the chain !?!?)
* interconnects ( at 300$ for an interconnect I can buy 3 amps)
* connectors, jacks, soldering joints, internal wiring
* interface impedance mismatch and changes ( e.g. by analog volume control)
* coupling caps (- these parts (a single cap) can do more bad to the sound than that what you'd
experience between 44.1 and 96 material)
* noise and phase shifts all over the place - as a result of above
* EMI /RFI - - as a result of above
* asf asf.
And then there's another very nice feature, which comes with "some" of those amps (if implemented by the manufacturer) :
The internal volume control is of analog nature. On the first glance this does not sound that special.
And it sounds kind of weired, when talking about a full digital amp.
However. The way it's done is the key issue. TI e.g. is doing it by changing the actual pulse output voltage ( the supply voltage of the output stage) of the amp. It's the last voltage source right in front of the speaker.
Doing it this way is extremely simple and most effective in comparison to all other types of volume control I'm aware of.
It IMO can't be done better than that.
Just to outline the volume control subject a bit: On the typical volume control setup in a traditional DAC/AMP setup, you reduce the pretty low level DAC output signal (usually 2V max at 100% VC position) - by estimated 18-24db. I think it is even more in most cases. Nobody I know runs the volume control at 100%. What's left after 24db attenuation is 0.125V at best. Basically you dragged your valuable 2V signal much closer to the noise floor.
That obviously seriously affects the signal-to-noise-ratio on your amp input.
The worst thing of all is that this seriously degraded signal gets amplified by at least 30db on the amp again.
You should know that the majority of amplifiers come with a fixed gain setting in the area of 24-30db.
Example: 4V on a 4R speaker gets you 1W for your speaker. Most speakers out there are as efficient as >89db SPL per 1W. You'll measure 89db in 1 m distance of your speaker. That might be an acceptable in-room listening level for the majority of people out there. 4V would be as much as you'd need.
The actual gain required would be in this case???
Yes. 6db only (instead of 30db that your amp delivers) on the 2V DAC output.
Nobody seems to care respectively to be doing anything about this serious problem you'll find in
almost every chain out there..
However. There is a solution.
Those full digital amps (some of them) come with an output voltage control which is not attenuating
the audio signal. These amps are adding gain "on demand" right in front of the speakers.
It's actually a flexible "gain" control. 99.9% of the volume controls are "attenuators". That's a huge difference.
The best possible place where a close to lossless volume control can be done is right in front of the speaker, the way it's done with some of those FDAs. Here we go.
All above advantages translate into crystal clear, very dynamic, very black and very neutral sound. You'll notice very low coloration artefacts, a wide and deep soundstage.
Some people feel strange about their listeing experience first.
Usually you'll hear comments like: the "body" and "warmth" is missing.
Yep. "Body" "Warmth" -- result of filter effects, colorations and phase shifts due to whatever problems in the chain.
I can tell you one thing. I'm more then happy to get rid off those nasties. The good thing about those flaws
which lead to "body" and "warmth" is that they als mask other problems in the chain. These might become appearent with a more resolving system.
The best of all - you'll get several of these amps at a fragment of cost of the usual "audiophile" DAC+AMP setups.
A small FDA amp - the actual chip - starts at 3$. Yep -- that's right. It just requires very littly surrounding electronics to make it sing.
As with other electronics, better and good quality parts and power supplies will generate better sound.
I'm sure the established "audiophile" industry is scared to death about these developments and this technology.
Advise: Stay away from overpriced x000$ FDA products. These are usually built around the same chips as used in 300$ amps.
What I experienced is that those FDAs don't need super clocks inside to perform pretty well already.
What they need are clean and fast powersupplies and you need to feed a nice low jitter and low noise digital signal.
There are USB/SPDIF interfaces out there which do a great job for less then 200$. These can be used to feed those FDAs. Latest FDAs come with USB interfaces at 80$!
The transport, a streaming device, a standalone PC or a smartphone/PAD feeding those amps
directly, will make a nice efficient and high quality chain. If you mate such a device with your FDA
you'll experience pretty high quality music presentation at a reasonable price.
Rank 1: NAD D7050 @ 850€ (available since 10/2013)
It's not only a FDA, it's also a sophisitcated streamer,
comes with async. USB input, SPDIF and Airplay and
even offers subwoofer outputs.
Did I mention that it sounds awesome!?!?
Rank 1: HifiMeDiy DDX320 (100$)
Connexelectronics SMPS300RS (60$)
A small list of vendors/products:
NuForce DDA-100 (549$) - NEW
Sumoh Amps (199/269€)
QLS QA100 (350$)
Pure Audio Stream Maestro 50 (270€) - UK
Wadia PowerDAC 151 (1300€)
NAD C390 DD / M2 (2400€/6K)
(Note: You might have heard of the giant killer DAC NAD M51. It's just a PCM/PWM converter without the amplifier power stage that you'll find in the FDAs.)
In July 2013 NAD will launch a new FDA product line.
These FDAs will be called D7050 (999$) D3020 (499$). These amps will offer asynch USB, Bluetooth APX-T , the big one even UPNP/DLNA network streaming and standard digital SPDIF inputs of course.)
Lyngdorf TDAI 2200 (2800€)
Kratos Core audio Technology (2500$)
Sony S-MASTER series ( Sony has been the first who came with multichannel FDA years back)
(please let me know if you're aware of more amps)
HifiMeDiy UD20 (80$) -- Asynchronous USB Full Digital Amp
-- just 20W
-- same technology as used in DDX320
-- potentially better output inductors
-- not really DIY
HifiMeDIY - DDX 320 V3 ( 100$!!)
-- that's what I use currently - replaced a Sabre DAC+AMP
-- SPDIF/Toslink inputs
V1 vs. V2 improvements
--better SPDIF input-chip WM8805 (50ps jitter)
--much better SW control options
--no 96khz resampling anymore
HifiMeDIY introduced an update again V.3
Better layout, traces, Panasonic FC Caps, pulsetransformers
Mini-DSP miniAMP (60$) (I2S only - mates great with the MiniDSP USB Streamer or WaveIO USB
A small list of integrated amps sometimes also called Power DACs.. These combine a DAC and an amp under the hood. These are NOT full digital amps!!!.
Teac A-H01 uses Burr Brown 5102 DAC and ICEPOWER 50asx2 (338$) or the newer AI-501DA (799)
which comes with an ABLETEC amp instead of ICEPOWER.
Sinewave G36D or similar G33D (295$)
Hifi Akademie PowerDac (1890€)
Wired4Sound mINT (1500$)
Patent - Technology used by STM ( DDX320)
DDX320 V2 - Modifications (Status 01/2013)
The HifiMeDIY DDX320 V2 @ 106$/82€ is my Full Digital Amp of choice for the time being.
The stock device performs quite nice. Though I wasn't really happy with the performance of that stock DDX. It was good, but not what I'd call audiophile.
Over a period of 4 weeks I spent some time to apply this and that modification.
Basically I applied those mods I'd try on all my audio stuff.
But first, before you start your journey, I highly recommend that you let the amp break in.
It really improves a lot over the first two weeks.
You'd have a real hard time to distinguish between break-in effects or tweak effects.
My Modification -Summary:
* Decoupling needs to be improved. E.g. on clock/VDD STA320/Toslink.
Decoupling caps can be soldered right to the pins or very close by.
There is no local decoupling close to the clock and Toslink. The STA320 digital and
analog supply pins need to be improved.
* All (just 6) rather low quality Elna Stargets should be swapped
The choice of small caps can be improved - IMO at least Silmic II can be used.
I still had some small Blackgates NP around, so I used those.
Extra cost are almost neglectable. For digital decoupling (e.g. clock and Toslink) , lowest
ESR Oscons SEPC are highly recommended.
* I swapped the 330uf Panasonic FC (used as 3.3V buffer) with Blackgate 1000uf NP.
* Output coils got swapped (e.g. shielded Wurth WE-PD). And best
mounted underneath the board. Note: I still use my Mundorf aircoils (you might have space
and radiation issues with these).
If not possible to swap coils the existing coils needs to be stabilized with e.g. BluTack or similar.
I use Henkel Teroson sealant for cars (kind of modeling clay)
The coil mounting is pretty poor - wires used as stands. These vibrate and ring like hell.
* Use of LipoFe4 3.3V battery supply instead of the Techcode DC1507 DC/DC converter + output
filter. I use 3 A123 cells in parallel.
Note: It's a bit tricky to desolder the DC7DC converter. That's why I cut off the legs of the chip.
But obviously that means you're entering the "way of NO return". ;)
I also removed all DC/DC converter related parts ( e.g. inductor). The batteries are hooked up right
to the 330uf Panasonic FC or in my case a large Blackgate.
This battery tweak is IMO one of the - if not - the most important tweak(s)!
For future board revs HifiMeDIY should consider an option for an internal/external 3.3V supply
* Lelon power caps need a bypass turbo. I used 0.52uF KP cap and a little silver-mica.
You also should stabilize the Lelons - with e.g. blue tack to avoid vibrations.
You might take off the plastic cover off those caps.
Note: I do have the impression that the Lelon's are not worse then e.g. Panasonic FC.
That's why I did not replace these caps.
* In case of 24V DC operation, the diode you'll find right at the suppply input can be shorted!
In the future HifiMeDIY should consider a 4th input for DC only.
* I use a Connexelectronics SMPS 3000RS set to 28V. 28V seems to be
the best voltage.
* For those who run SPDIF. Get yourself a pulsetransformer ( e.g. from Pulse - Pulse PE-65612NL -
IMO the best out of 10 I tried (that inlc. stuff like Scientific Conversion transformers)) to isolate
Though --- avoid double isolation! Check if your source/transport comes with pulsetransformer
Unfortunately HifiMeDIY didn't follow my earlier advise on this one. There are plenty of inputs.
At least one should have been transformer-isolated
* On the SW side I turned the
- bass section off and
- activated automatic display off and
- highpass and all EQs off
* The DDX320 resamples everything to 24/96!!!!
It excepts everything from 44.1 to 192, but resamples everything!!!
I do highly recommend to do the resampling on the transport/PC side.
Note: There is no AES/EBU input. And it's not possible to connect via I2S! The MCU
expects a sample rate notification from the WM8805. That's not working if
I2S would be used directly.
Note: On the DDX320 V1 I was using the crossover function ( highpass) at 80Hz.
That lifted off some low frequency stuff from my fullrange speakers. I liked what I heard.
With the tweaked V2 in place, I can clearly hear the DSP related losses.
That's why I run the the V2 without highpass nowadays.
Note: All caps (piggy-back style) and coils are mounted underneath. The board is mounted on stands.
As you can see I put some modeling clay (Teroson sealant) on Caps, coils, Toslink and clock.
As you can see above, most of my modifications relate to the power supply, such as cap swapping, decoupling, battery.
These mods should be managable by everyone, who's done some basic modding.
The key modifications are IMO the 3.3V battery supply and the output coil replacement.
Still, I'd highly recommend you'd do them all. You might use your own parts of choice.
Please let me know if you find even better mods or let me know your parts of choice.
Bottom line. All above tweaks cost you (and HifiMeDIY in case of a little re-design) close to nothing.
With those mods being implemented, you'll end up with a pretty high-end full digital amp.
As I said before. My TP Buffalo (LiPoFe4 powered+ more tweaks) + amp chain is settling dust.
I'd like to say that even without all these tweaks the stock DDX320 V2 is a nice sounding all-in-one solution. With the tweaks in place it's an outstanding value and value for the money.
Who needs more then that???
Just to mention it. I'm not using the amp for < 100Hz. And my Bastanis speaker runs at an 98db/W SPL. Driving low frequencies and/or low sensivity speakers might show the limits of this little DDX320 amp.
My tweaked SB Touch connects via Toslink (SilfLex) to the DDX and SPDIF to a Behringer DCX2496 for the subwoofers
(Note: Make sure you decouple the SB Touch Toslink Transmitter with e.g. 220uf low ESR Oscon SEPC right at the Toslink transmitter pins - that's must if you go Toslink!).
Make sure you have a clean and fast 24V supply and a very good transport feeding that amp. (I'm running buffered low ESR batteries)
@HifiMeDiy: I'm not a board designer, though when looking at the board the routing of powerlines, the output stage and the positioning of decoupling caps can and should be improved - even if the board gets a
I also think that the output section needs some more air to breathe and better coils.
But that's someting for HifiMeDiy to consider for V3 - if that will ever come.
You might check out the QLS QA100 layout. ;) Obviusoly I can't tell how that one sounds-