Re-direction Part 3 - First Verification

This will be a long entry. Last weekend I was back home for a few days, and I took the opportunity to set up my speaker system. The main purpose of this was to verify two things - listening and volume and bi-pole speakers. The set-up was quite simple and there was no effort to optimise cabling, speaker position, etc. The amps were my previously completed LM3886 chip amp and the second was the Decware Zen kit. Both of these amps have been completed for over 3 years, but have had very little chance to run due to life circumstances. The speakers were a 'closed' bi-pole FE-127e towers.

A few notes about the amps. The Chipamp was made from the stock with no upgrades or modifications of any kind, had it was not a dual mono. The Zen was significantly upgraded from the kit. All resistors and most capacitors were changed. It has been modified to use a 5U4G rectifier and has a large 40uF ASC oil cap. The tubes used were all Russian 'EV' grade 6n1p and 6p15p.

The first arrangement was the Chipamp with the speakers set up as dipole. That is, both of them moving in the same direction (even though they are mounted back to back). This would be akin to an Open Baffle arrangement, as the dispersion pattern is a figure 8. Previously, I preferred this arrangement, and my starting point was from here. The source used was by QA350 running on battery power and with a range of CD ripped or HD Tracks converted WAV files.

First time, everything worked. Which was quite a relief. I will now talk about the Chipamp and Dipole in a whole. The sound field was strange, and with the nulling effect at 90 degrees of the speakers was quite prevalent. The sweet spot was indeed a spot, both vertically and horizontally. The speakers are mounted a little lower, around 800mm off the ground, and was lower than my ear level. I found that I had to be pretty level with the speakers and in the middle for the sound to be good. But when I was in the right spot the sound was pretty good. After listening to a few familiar tracks, I realised that this particular combination didn't sound right to my ears. So I decided to switch it to a bi-pole configuration.

In a bi-pole configuration, the speakers are wired so that they both move in the same direction relative to the speaker baffle face that they are mounted on. That is, they both move in to the speaker and out of the speaker at the same time. The net effect of this, was a dispersion pattern that is very omni-directional. This was quite obvious when standing 90 degrees to the speakers. Unlike the dipole, there was no reduction in volume.

Sitting back and listening, I found two things. Firstly, the sweet spot is much larger and more forgiving. I could imagine a few people being in that spot. Secondly, the sound stage and sound field was much larger and there was more immersion. After a little more listening, I quickly realised that this was the sound that I preferred.

After I got to this point with the speakers, I decided it was time to try the tube amp. I cranked it up with the Chipamp and did some casual measurement on my iPhone. The levels at the listening position was around the 80 dB marked, and hovered around there with peaks just shy of 90 dB. At this point, I realised that my goal of 100 dB plus at the listening position is not required for me. Therefore, I have to revised my design criteria, and it seems that I would like the amp and speaker combination to have total capacity to reach somewhere in the 90 dB range at the listening position. At this point, I am thinking specially between 96 and 96 dB. This would give some additional headroom, and would significantly open up the speaker and amp combinations.

The spec of the FE127e is 91 dB, and in a bi-pole configuration, it should be around 94 dB, which, in newly revised, should only need a 2-4W amp to reach the desired levels. I did notice that listening at loud levels, I did get a slight headache, and it was fatiguing. Hence my next move to change the amp.

Plugging in the Zen did not alter the sound dramatically. However, having only listened to these in the span of a few days, I imagine it'd be difficult to find all the differences. However there were two important differences that I noticed with the tubes. One is that it was not fatiguing at all at any levels. Second, the sound was more full, whereas the Chipamp was more sterile. So in this short sessions, I have decided that Chipamp (and solid state) was not the way for me moving forward.  Once I settled on this, I spent probably 6 to 8 hours in total listening to this setup. And it was really a delight.

Overall, the sonic presentation was natural, easy and spacious. The bass was not loud, but was present and without authority. This is one area that is becoming more and more important. That I need to really find a good speaker to supplement to provide the missing bass presence. With the Zen, two observations was made. The first, is that almost at full volume, the levels were around the very low 90 dB, maybe hitting 92 dB. This was probably as loud as I would want to listen to for a period of time (more than peaks or more than a track). Secondly, I found that classical music, which to to date, has been a little disappointing in my other setups, was very pleasurable. It didn't seem constrained and didn't feel like it was running out of steam.

So, the little effort that I made this trip provide to be very valuable. The following items have been confirmed.

1. Peak levels would be in the low 90 dB range, and thus total system response should be at least 96-98 dB at the listening position.
2. I prefer bi-poles to dipole. I didn't get to chance to wire it as a monopole - though I was thinking about, but I was just enjoying the sound so darn much.
3. I prefer the Zen to the Chipamp. And thus, I could make the extrapolation that I prefer tubes to solid state sound, or Class A to Class D. 

This has been both important and enjoyable, and now I have a few months to prepare my plan of attack before I am back at home permanently.

Re-direction Part 2 - Speakers

Described by many as the heart of the system. Finding and/or designing the right speaker is probably the most important, as the approach I am taking is from the ears back. There is so much discussion and research on speaker design, and most of which is well beyond my capability to completed comprehend. However, from my research, there are number of key issues in obtaining the best quality speaker seems to be.

Crossover points - avoid if possible 250 Hz to 4 kHz, about 4 octave range. This is the most critical range of musical information and many describes it as the most critical point to avoid any crossover.
Use of neodymium or Alnico magnets seem to lead to higher quality of sound.
Use of cone materials that prevent breakup well outside (i.e., 2 octaves or more) the frequency range being played on the speaker.
Different speakers were designed to different enclosures, make sure the enclosure fits the speaker. General rules of thumb like using high Qts on open baffles, etc.
Important to have a smooth frequency response with good dispersion.

If we take the above first two items in our next design assumptions, we would be left with three speaker system. One for the LF work under 250 Hz, and one for the mid-range work from 250 Hz, and a HF unit crossing between 5kHz to 10 kHz that would take it to 40 kHz. The choices for the various drivers would be limited in the MR and HF areas. But once we take into consideration all the criteria, the range of speakers that can service in our requirements become somewhat limited.

However, in recent review of components and equipment, I have realised that I have amassed a range of speakers. FE207e, Lowther DX3, Saba green cones, Beyma Coax, Joran JX92s, FE127e, FT17H, and Alpha 15A. To be honest, to get good speakers cost money. And living in Australia, the cost of shipping of most speakers are quite sizeable. So the vein of being economically responsible, I will have to design a speaker system that uses one or more of the existing speakers in my collection. It appears from what I have, that it is mainly full range speakers, with the exception of the Beyma Coax. All the other speakers will mate well with a sub crossed at below 300 Hz or so, and can all probably do with a super tweeter (Fostex FT17H).

Therefore, it leads me to designing two speaker systems. One, the full range with sub and super compliment, and the other around the Beyma Coax. My next post will deal with the design of the Beyma Coax as a possible contender.

Verification of thought process

Since I will have a short trip home in the coming weeks, it is my plan to verify my second assumption, that is the 104 dB levels. My goal is to take out my pair of closed FE127e towers (yes, those with the front dust cap crumpled) and play them using my Decware Zen and then LM3886 Chipamp in the 100 cubic meter room. To hopefully negate the effects of the crumpled dust caps (yes, they are both now crumpled…) I will rotate them around and use the 'back speakers'. Wonderful thing about dipole and bipoles! The towers are made from 19mm thick Tasmanian Oak and have light filling. But the top and bottom are loose, leaving plenty of opportunity for tweaking.

The drivers are all terminated separately so that I can experiment with dipole, dipole and the parallel and series. I will listen to my favourite music, and see what levels are acceptable. If, and I stress if, they are suitable with the Zen, then I would see so many options opening up in both speaker and amp combos. If they are no loud enough with the LM3886, then I really need bigger amps and/or higher sensitivity speakers.

Through this, I will also experiment (hopefully) with the monopole, dipole and bipole arrangement. Which should give some further guidance as to my preferred arrangement. The source will be my QLS350 straight into the amps. I might try bringing in my HT sub, to see the effects of low end reinforcement. I will consider this my most important experiment. Speaker configuration, low-end reinforcement and power level combinations.

Note - to be honest, I am keen to see if I can get these to work well with the Zen, and then I can upgrade them to the F120A or even the new Jordan full rangers!

Re-direction Part 1

After much pondering about my audio system, especially on my imminent return home, I have been through many of the same processes as I have done in the years prior to my departure from home. At the end of the day, it always came back to the same things as I have covered in my previous blog entries. Namely, back and forth between an all tube amp system or a tube plus SS system (for more power) and that of the speaker system. Which has oscillate between a high efficient system using horns, or a lesser (but still quite) efficient system using a traditional full range speaker driver.

Thus, I have come to a different process of deciding. A little background about my gear generally. In all my interests and hobbies, I plan and want to just have one - what is perceived by me as the ultimate, and then that is it. Not that it has to be the most expensive or best, but something in my mind that is no compromise. That is case for my cameras, computers, watches, cars, bass guitar, etc.. I have found that going around from low end to high end, ends up costing more money and time. But in the audio field, as I started with very little knowledge, and wanted the DIY route, it was natural that I take a longer journey.

In terms of headphones, I have I have almost reached that point. With my (soon to be built) C3g amp with HD650. Even with my existing range of built amps (about 6 in total) and headphones (over 4 high-quality pairs), I am more than a happy camper.

With speakers however, it has been a (very) long journey. Mainly due to moving houses, moving country and having children. But right now, I am in a better spot to move towards that end goal. So, instead of choosing an amp and speaker, I am going backwards, and choosing what I want in terms of sound, and then working back the audio chain. That is, desired sound, sound levels, speaker, amp and then source.

What I want is a system that is able to reproduce audio between 20 Hz to 40 kHz with high fidelity and ability to engage me beyond what can be 'objectively measured' - that is, it gives me enjoyment. What would I mean by high fidelity, I would assume low distortion and the frequency response to be generally smooth to within 3 to 6 dB in the frequency range, with a drop of say -6dB at 20Hz being acceptable.

Being honest, my listening room will never ever be larger than 7.0m by 5.3m by 2.7m high, so about 100 cubic meters. But at the moment, the listening room is 4.7m by 4.0m by 2.7m, about half the volume at 50 cubic meters. The system should be built for those two rooms.

The next question for me was how loud? I have been looking around the Internet, and measuring the values that I typically listen to. It seems that a value (for peak levels) of around 103 to 109 dB was about right, with most program levels being around 80 dB to 90 dB at the loudest normal levels.

In the larger room scenario, the listening position would be about 4m from the speaker, and the smaller room, it'd be 3m from the speaker. Which is not all that different. From this information, I would then determine the appropriate levels of speaker and amp performance to meet those dB levels. A listening position of 4m is about a 6 dB loss and at 3m, it'd be about 5 dB. In either case, the differences are marginal, so let's assume 6 dB loss due to speaker distance.

At this point, I will ignore room gain and other incidental losses, as they compound the complexity. For peak levels, let's assume for simplicity sake 104 dB. Therefore, the entire system of amp and speaker should be able to hit 110 dB. Therefore the first assumptions that are quantifiable are:

1. Frequency Response range of 20 Hz to 40 kHz. -6dB at 20 Hz and within 3 to 6 dB over the range (that is plus or minus 1.5 to 3.0 dB).
2. Combined maximum SPL levels of system should be 110 dB at 1 W at 1 m.

With the above in mind, we can now start to tinker with combinations of speakers and amps that meet the above requirements. The first for me is the philosophical combination of the two devices. So, if the speakers were to be 101 dB efficient, the amp would need to be at least 8 W in power. If the speakers were 92 dB, the amp would need to be at least 64 W in power. So, as the speaker efficient increases, the amp options open up for low powered tube amps. And as the speakers become less efficient, the ability to use tube amps rapidly decrease. The table below summarises the combinations.

Speaker Sensitivity // Amp Sensitivity
110 dB // 1 W
107 dB // 2 W (45 Amp)
104 dB // 4 W (2A3, AD1 Amp)
101 dB // 8 W (300B, KT88 SE, F2a Amp)
98 dB // 16 W (6C33, 211, 845, GM70)
95 dB // 32 W (211, 845, GM70)
92 dB // 64 W
89 dB // 128 W

Given my general interest in using Class A Tube amps, where generally SE tube amps limited to below 16 W or so, and PP tube amps around 32 W for Class A, it would appear that unless the speakers were to be at least 95 dB efficient, the use of tube amps would be not possible. Even with the rough limits mentioned above, the voltage and/or current requirements are quite large, limiting the ease of DIY. However, the choice in the number of amplifiers used has not yet been addressed.

Therefore, if I want to use tube amps (and I do) then I would have to find highly efficient speakers. This would generally rule out most of the speakers available. Focusing only on the mid-range (or pseudo full range) would limit the choice to high efficiency speakers such as 'traditional' full range family (e.g., Lowthers, Fostex, Supravox, etc.), horn loaded compression drivers, and other very efficient mid-range (generally) PA speakers.

Well, a longish entry, Part 2 of this will cover the issues dealing with speaker selection.

Relaxed session

Well, I had the first session in a long time with my EM4 in a somewhat high quality arrangement. I was using the O2 headphone amp and the D-zero DAC from my Macbook, using Fidelia as the source. Short commentary.

Source does matter. Some files were better, and some were worse. But I must say that hearing Bach's Cello suites through this setup was very intimate. In the stillness of the afternoon, and using IEM, I really felt like I was in the room with Janos as he plays the cello suites. On another thought, 1812 Overture was just flat, even on 176.2 kHz source file. I think that complex music just doesn't play well with the setup. Not sure why, but it was just not as inspiring as other kinds of music.

But, it was a refreshing and enjoyable session of one and a half hours.

EL84 PP Idea

Well, I was on the train home and was thinking about the Beyma. I was considering have it passive crossover. Using copper foil inductors and Jantzen caps, it turned out to be around US$350. I was then thinking what if I just built two simple EL84 PP amps instead, and then I can experiment with a cheapie PA active crossover. This was, I can tune them and then once the sweet spot found, I can do a hardwired active crossover. Would that be cheaper?

Anyway, more incoherent thoughts later, see below. 2.6W in SE mode, and thus is Class A PP, it'd be 5.2W. 2nd harmonic should cancel out (mostly) and the 3rd harmonic should sum, but the predicted 3rd harmonic is pretty low. The PSU below using a choke input. It could be a simple, 'cheaper' build. But it could also be quite good. The output transformer is 8k in SE operation.

Oarta output transformer

It looks like the Lundahl LL1623 rated at 90mA would be the best choice for this amp. The amorphous core price is less than half of the Tamura, but more importantly, it is available. The interstage transformer would be the next choice. This would probably be the most important choice, given the different voltage swings that needs to be accommodated by the different tubes.

I think at this point, it would be important to at least try to limit the range of tubes to be used. This has to be correlated with my current stash of tubes. In this, I would say that there are four families that I would like to run:

1. 45, 145, 245 and 345. I have a number of pairs and have been universally hailed as an excellent audio tube.
2. F2a. I have three pairs of these long life tubes.
3. KT88/EL34 family. I have two quartets of the KT88 and EL 34.
4. 300B. I have one pair. 

Looking at the above, it would make most sense that I provide more effort towards the first three families given the volume of tubes on hand. With the 300B narrowed out, the voltage swings required are now limited to 30 Vpp to 60 Vpp, a little more sensible. This would mean that a tube with a gain of 20 would be plenty. At this point the 5687 and C3g seems the right direction. If I use a higher gain tube, I can use a step down interstage, which should be a good think overall. The other option, is to allow a different tap on the interstage to be used to connect to the grid of the output tube, which can select between say 1:1 and 2:1. But this may affect the performance and bandwidth of the interstage.

More questions... 

Oarta Output Transformer

It seems that the Tamura F5003 is in limited supply, and may not be available. So I may have to revert to another brand of amorphous core. Maybe Lundahl?

Oarta PSU

One interesting point is that if the PSU is cap loaded with a 10uF, at 200mA, it will give around 380V. But if I remove the cap, and have it choke loaded (using the LL1673), at at loads of around 100mA to 150mA, it will give voltages of 260-280V. Perfect for 2A3 and 45. So in this case, if I can provide switchable PSU filter, I can further extend the range of usable tubes, rather than using resistors to drop voltages.

Oarta - Some Design Images

PSU Design.

KT88. 400R cathode 

300B. 740R cathode

KT66. 300R cathode.

EL34. 400R cathode - can share the same plate as KT88.

6550. 400R cathode - can share the same plate as KT88.

The Oarta - One Amp to Rule Them All

It has certainly been a quite a long time since I have done ANY DIY audio stuff. I blame this due to my relocation to Hong Kong for the last 18 months and having young kids. There just ins't time. Well, with the imminence of return back home on the horizon, I have started to dream up what I will finish up. In this long time away, I have realised that I like variety. But I don't want to build too many amps. So, I have taken a few ideas, based around a SET and merged it into one amp. Which I call - Oarta, or One Amp to Rule Them All, not exactly creative.

The basic premise of this amp is that the PSU will allow the supply of around 390V at 200mA max, with heater windings two sets of heater windings, 2x 6.3V at 2A and 1 or 2 5.0V at 2. This should allow me to use a large range of tubes that I have on hand: F2a, 300B, KT88 and EL34. And also open up a whole range of other tubes that can run at those voltages.

The second part is a universal output transformer. I have chosen a 5k tap, but may move to a 3.5k for more power. For this amp, it will be no compromise, so a amorphous Tamura F5003 or F5002 will be on order. The driver stage will be a medium mu (think around 20) that is interstage connected to the output. To accommodate the wide range of output tubes, I will probably want a higher gain stage, so C3g triode connected may well be the final choice.

The trick here is to allow the switching of the output tube. To this, I am thinking of an adaptor plate that allows the socket and cathode resistor to the changed. The bypass resistor will then be mounted on the chassis. In this way, I can have a large film cap that isn't changed with each socket. On each adaptor plate will be 5 connections; grid, plate, cathode, heater + and heater -. I need to find safe and reliable connectors that allow me to change.

The final trick is have a 6.3v and 5.0v winding, and finding a method to accommodate a range of heater suppliers. For the indirectly heated 6.3v tubes, this is no issue, but with the DHT tubes, like the 300B, PX25, etc., this will need some more work. There may need to be an allowance for something like the Tentlabs filament supply, that can run from the 6.3v to power a 5.0v and run from the 5.0v to power a 4.0v tube.

If this can be made to work, I have an amp, Oarta, that will allow me to change tubes, without having to change anything else. This will allow me to not skimp on the rest of the components, knowing that it will be universally applicable. At the moment, my current thoughts for the design are:

PSU.
Amplimo 7N1474 Transformer (here - I may need to supplement with additional heater taps)
2x 6AU4GT diodes
Cap input 10uF ASC Blue line
Two LC stages with Lundahl LL1673 200mA (here) each with ASC 50uF.
I am thinking about a separate LC stage (15H with 40uf) for each channel and an additional LC stage for the driver.

The main issue with the PSU is finding space for all the ASC oil caps and the chokes.

This will give me around 386V of B+ at 190mA current draw (simulating with PSU Designer). At this B+, the following would be the power output with different tubes (from SE Amp CAD) using F5003 all running at 80mA per tube.

300B: 7.6W with 120 Vpp
KT88:  6.8W with 62 Vpp
EL34: 8.8W with 60 Vpp
KT66: 4.5W with 46 Vpp
F2A: 5.8 W with 32 Vpp (by hand)

And I would think that this would also work with tubes like GU50, PX25 and some of the other more rare European or exotic NOS tubes. And if I don't mind burning off voltage as heat, it could be used on 2A3, 45,  AD1, etc. - again, options. Beyond these, there are still a number of issues to work through, but in my mind, the concept already has been resolved. I am now thinking about cathode bias for the driver, and most likely will head down the LED route. Given the C3g triode seems to work well with around 2.0-2.5v bias. 170V at the plate, 2.5v bias 15mA. Nice and simple.

The main course of action now, is to determine how best to build the adaptor plate, this will make or break the design. It must be safe, easy and high quality to preserve the rest of the amp design. And the last question is how to deal with the different voltage swings required by the various tubes. Which differ by a factor of 4. I do not want to simply waste it by using an attenuator, and the first thought was to have taps on the interstage, which would allow me to choose the right ratio, 1:1, 2:1 or 4:1. This I think is possible with the Sowter 9525. But I probably want to avoid changing the IT ratios as that will change the frequency response characteristics.

Or I could just vary my source output - say between +0 dBV, +6 dBV and +12 dBV. (Which incidentally is possible with the Wadia 121 DAC that I am considering, see manual). Or, I could just accept it, and have to set the volume levels differently for each tube. This though, needs more thought and consideration.

Minor Updates

Well, it has been some time since my last post. Again, work, family and travel taking me away from audio and music. I am currently enjoying my mobile setup of iPhone 4 and my Earsonics EM4. At home, I am continuing to enjoy music through the computer to the Li Te DAC-AH and the through to the O2 amp. A lovely combination.

A few notes regarding the O2. I cut one of the gain resistors to reduce the gain to unity. I has allowed me to use more the volume control. Before, it was so sensitive there was horrible channel mismatch. But now, I can use it at 9 o'clock. Still not great, but better. Also the volume control on iTune is at 20%. Again, not great. My next task is to reduce the IV resistors and the gain of the opamp on the DAC to reduce gain even further. My aim is to have iTunes/Fidelio at 100% volume, and have sensible levels at my headphones at the 12pm/1pm position.

Have been thinking about the DAC, and how to build one with deliberately low volume, and the mating it direct to a LM based amp (something like 'The Wire' or TPA's Ventus. But have been designing that (or drawing that) and is taking a long time. Also not looking forward to the SMD parts soldering. Not sure when I will have the chance to put it all together, as my soldering gear is back home...

Anyway, not much to report on. Hopefully I will be able to do a few things like publish my 'design' (I use this very roughly cause I am really cutting and pasting...) on the TDA1543 with the Wire. I plan to use the O2 power stage (giving on battery +/-8V ish and on AC +/-12V ish) coupled to a standard TPA SPDIF receiver setup.

QA350 and O2

Had a chance to connect up the O2 with the QA350. Having spent two or three sessions with the QA350, it has proven to be a very useful addition. From this, I can now take my own music and use to demo new equipment. For example, been planning on trying out the V200 with my EM4, and now, with the QA350, I can in someway have a more 'controlled' test setup - rather than using the vendor's setup. Secondly, this unit; whilst compact and light weight, is not really body wearable. Even with a jacket or suit, the size is just too large. With a bag, there is not a problem. But issues arise when you are walking around, with a delicate and shortish headphone lead, not the safest approach.

The volume control on both the QA350 and the O2 needs to be no more than a 10 degree rotation from their 'zero' position before the volume gets too loud. Not a good arrangement. I will have to find someway or padding the output volume down, or reducing the gain. This is why the V200 would be a good option, as there are DIP switches on the back to adjust pre-gain. Mated with their V800; which also has output gain selection, could tailor a solution where the EM4 can be used at sensible volume control positions - of which I deem as around 40%-60% of full rotation.

Currently listening to HD Track's 1812 Overture, downloaded as 176.4/24 converted to WAV via XLD, and then down samples by Sample Manager (using isotope Resampler set at 'High' and MBIT+ dithering) to 44.1/16. Just finished it on the O2, and to me, the sound was a little confined, in my head, and potentially a little flat. Having started with the QA350, I can't tell that much difference from first impressions. On a side note, both the O2 and the QA350 have had relatively little burning in time; probably at most 10 hours each. Whereas my EM4 have now got a very healthy 200 plus hours. The volume control on the QA350 has channel imbalance at low levels, important for me cause the EM4 are very listenable at those levels. Also the optical output light is always on, so there is a constant red dot on the main side panel. When the charger is working, I can detect a high pitched noise. Owning to the warnings about Lithium polymer batteries, I am keeping a close watch on them.

Back to the QA350. I dare to say, that with this very casual session, there isn't a lot between them. I think I could hear a few more details with the O2 then with the QA350; and the cannons were more powerful with the O2. In some way, you could feel the cannons - not the full body sensation, but just around the ears, if that makes sense... Both the sound stage is limited to between the ears. I know the EM4 can project sound around the ears (for me anyway).

Now moving onto the Computer setup. This time, O2 front end, from the Calyx, running Fidelia with the full resolution FLAC file. Immediately, the clarity returns! Soundstage is wider - straight away! The setup in Fidelia - 24 bit MBIT+ dither, device sample rate set tup 96kHz with isotope resampler set at 'High'. At this stage, I am running the volume at 100% on Fidelia and using the O2 to control. Changing Fidelia to point at '9 o'clock', has shifted the O2 to be pointing at '10 o'clock'. Differences in sound? A little loss in clarity, but very minor. With dither turned off, and the Fidelia set to 9, I could hear a difference, or missing music, that was there before, when dither is on. Overall, the presentation on this setup is more musical than through the QA350. Oh yeah, the cannons on this were way more body, substance and 'cannon-like'. Night and day. Through the QA350, they just sounded like pop guns, or cannons in the far distance.

The last test is to play the same 44.1/16 WAV file on the QA350 through Fidelia. Oh yeah, the file size is 524.4 Mb compared with 150.9 Mb. Interesting in that the jump from 176.4/24 to 44.1/16 should be a 6 times reduction in information. So, given FLAC has some compression, the full size 176.4 file should, by rights be 905.5 Mb, therefore, the FLAC file has losslessly reduced this by around 40%. But this comparison with the higher res file is moot - the Calyx is limited at 96kHz.

Firstly, the sound stage, again, much wider. The rendering of the music, is more lively and more musical than the QA350. There is a loss in clarity between the higher res file and the lower res. A little more separation in the music and instruments. It is, 'more focused' in photographic terms. But the difference between the QA350 and Fidelia is quite noticeable. I would vote Fidelia in this instance, based on this one track. Again, the cannons on this war much more powerful with more body and substance.

Finally, the same 44.1/16 WAV track through iTunes. Smaller soundstage - less clarity and detail. Very similar to the QA350. If anything, it sounds a little more confused, or muddy, than with Fidelia. What it does sound like, compared with Fidelia, is that there is a limit to the music, in both sound stage, dynamics and clarity. Fidelia sounds like these restraints have been removed. Turning up the volume does not restore, or give iTunes the same energy as it does on Fidelia. The canons are definitely more like the QA350.

After this track, Adele's 'Daydreamer' came up quickly, and it sounds pretty good on iTunes. The guitar strumming and her vocals, focused and articulate. Moving quickly to Fidelia with up sampling to 96kHz, improvements in the sound; but the extent mot as great as that with the 1812 Overture.

Now, what does this short hour session tell me;

• I can hear differences between the same track (source at 176.4/24) played between 44.1/16 and 96/24.
• Dithering on Fidelia is required when using it as the volume control.
• The headphone amp sections on the O2 and the QA350 is in principle very similar - but sure to be differences that can be told over a longer, more rigorous session. 
• There wasn't a huge difference between playing from the SD card and the computer. Perhaps this whole jitter issue, etc, is the storm in a tea-cup? 
• The soundstage, clarity and body is much better on my Fidelia setup than the QA350. 
• QA350 hasn't been burned in. 
• Cannon love on 44.1/16; best Fidelia, then by a long margin iTunes, and close behind that QA350. 

What I cannot conclude;

• QA350 sounds inferior to the computer. Using different files and sampling is surely more of a difference.
• That my opinions or observations can be repeated. This is not a double blind A/B. This is me on a quite afternoon listening to the same track on 5 different setups. Not switching back and forth between each one. Each one in its entirety and in full, followed by the next. So take the above with a grain of salt. Also, this was done with one track, a classical track. At the end of the day, YMMV, and so will mine, from day to day.

QLS QA350 Mod V2

Just purchased the QLS Mod V2 - to try out some decent portable audio in contrast (and comparison) to my iPhone 4. The first thing I noted was the supplied SD Card (A Sandisk Class 2 2 GB card) did not work with the player. I had to use my Class 10 Sandisk - this works fine. At the same time, I have also tried to convert my files into WAV. For this, I am currently using XLD to convert to WAV format, and then Audiofile's Sample Manager to down-sample to 44.1 kHz and 16 bit. A little bit convoluted, but still, it works.

At the moment, the first track, after 0 hours burning in was Holst's Mars from HD Tracks. I have only had a very brief time with this, and a few quick notes when listening to a more familiar CD, Coldplay's X&Y. The interface is refreshingly simple. Some say too simple, but I think its a great minimalist approach. I have no issues with that. It is quite a nice departure from a full blown computer playback - absolutely no frills.

When I am listening to music, I am not too concerned about the CD cover, track name, etc. I just want the music. So far, there isn't much hiss or noise when using the EM4. Which is a good sign. Increasingly, I am of the feeling that at 59R the EM4 aren't as susceptible to noise as I thought that they would be. Which is a good thing, but also, I am of the feeling that they probably need more voltage to drive well, as opposed to being current hungry. So perhaps they can be seen more like my HD650 in terms of amplification.

Anyway, back to the QA350. With the Mod V2, this had apparently an upgraded headphone section; using AD8397 as the driver. I must say, that at this stage, the sound is quite enjoyable and articulate. Nothing overly wrong, or excessive at all. The bass is probably a little loose, but the mids and highs quite well defined and controlled. It lacks a little bit of life and energy, especially on complex parts, it feels like it is losing some control over the music. But in simple guitar passages - quite lovely. Bear in mind, this unit has had zero play time thus far; so all comments need to be seen in this light.

Doing a quick comparison with the D-Zero over one track - 'Hardest Part'. The sound is a little more lively and wider on the QA350. The bass is more 'focused' and a little softer in the mix than with the QA350. This was played back through Fidelia, with volume control all the way up, and volume controlled from the D-zero. The vocals on the D-zero has a slightly duller and small image. The music in general was more lively than the QA350. In very short, brief and un-critical manner, the QA350 had a more musical balance, that already feels more enjoyable. But then, the problem (?), maybe there are just so many variables with computer playback, that it's hard to figure out just WHAT is the source of the negatives and positives.

Calyx Coffee - The Inside Story

On a roll, here are the innards of the Calyx Coffee. It appears from my causal observation that the headphone output is not driven by its own amp circuit, and is paralleled with the RCA outputs; which I presume is coupled by the Fine Gold caps directly to the ESS9023's output.

1. ESS9023P - DAC
2. Tenor TE7022L - USB Receiver
3. SC12.000 A1026 12 MHz clock
4. LD1117A - Adjustable regulator
5. Atmel ATMH038 - Some sort of Microprocessor to control the buttons
6. Nichicon Fine Gold 220uF/35V (x2)

iBasso D-Zero - The Inside Story

Finally (!) had the chance to open up the iBasso D-Zero and have a peek on the inside. Enjoy. Some of the major components;

1. WM8740SEDS - DAC Chipset.
2. AD8656 - Dual CMOS Amplifier
3. PCM2706 - USB Receiver
4. HEF4011BT - Input NAND Gate
5. NEC UD2-4 - Relay
6. Nichicon FW(M) 470uF 16V (x4) and 100uF 10V (x2) Capacitors
7. KDS 12.000 MHz Clock of some sort.

O2 Amp and the EM4

I finally (!) got around to purchasing a 3.5mm to RCA cable so that I can now connect the Calyx Coffee to the O2 amp from JDS Labs. Also of note, was the trial installation of Amarra Mini. Last night and right now, I am listening to music stored on an external firewire drive connected to my Macbook Pro. The music is running through iTunes/Amarra and the out to the Calyx and then O2, finally to the EM4. Currently listening to Adele '19'. Last night, I did a quick A/B between 16/44 and 24/96 on REM's 'Losing My Religion', as well as some casual listening with my favourite tracks.

Back to Adele, the first track, I can only describe the guitars as one of the most realistic I have heard through my headphones before. The articulation and intonation of the guitar, along with the strumming and 'shimmer' sounds very real. Having limited experience with both orchestral performance and guitar/drum band gives me some basic insight into how things sound when played live.

From this, and with the current setup, I can say that the bass and guitars have a very clear definition and sound to it. Comparing this to the other tracks, I can say two things about guitars and the EM4s. Acoustic guitar sounds phenomenal; warm, crisp and articulate. Bass guitar sounds just like on stage through - compared to my own bass playing experience. Mind you, this O2 has NO burning it and hasn't had its battery charged since it was delivered. Again, due to my laziness, I have not yet got an AC adaptor...

Again, with the current setup, I am convinced that the headphones (and by this assumption, speakers) play the largest part in the tone and sound signature of the music. Having dabbled very briefly into 24/96, I can say that, the resolution has the next most important effect. I am not not 100% sure of how and to what extent the Transport, DAC and Amp affects the music. But I can say that through the iPhone and this current setup, there are differences. But surely not as much as changing headphones. If I had to break it down, I would say that for now it'd be 50% headphones, 25% source and 25% transport/DAC/Amp. Perhaps this will change over time, but it will need far more exploration.

Well, the next few tasks are to decide between Audirvana and Amarra; and then find the right DAC setup.

A few quick thoughts regarding the DAC setup, after long thought, I am really looking for a compact solution that is airline luggage suitable. That rules out 19" wide units, and really limits this to one or two boxes that are at least half rack width. Not knowing how much travel is installed for the future, I would like to have a relatively portable system. Perhaps an A and B system, where A is the more permanent one, and the B is the one that I can take on short trips. As much as possible, I would like to share components, as I am trying hard to simplify my setup. So thinking more and more, the Halide Bridge or DAC HD makes so much sense, it removes cables and power supplies. It just makes sense. But the question really is going to be, will they deliver an improvement and get me closer to the music and quicker to relaxation. The search continues.

Q&A on Violectric V200 and V800

Another Q&A on the Violectric V200 and the V800 DAC for use on IEMs. 


1. For the USB input on the V800, is limited to 24/96kHz. It appears to be an add-on daughter card. In the future will you be releasing a 24/192kHz USB reciever card, and thus, can that be installed as an upgrade to the V800?
[VE] You are right, the USB input is sdituated on a daughter board and may be upgraded.
We intend to refurbish this PCB when a working 24/192 solution is deliverable. Currently it is not.
This new card will also be offered as an upgrade for existing systems.

2. Is the volume control in the V800 using the attenuation within the DAC chip? Or does it attenuate the signal after it has been processed? Where would be the best place to use the volume control? If the V800 is indeed a digital volume, then I'd assum that it'd be best to apply it there at the V800 and leave the V200 fully open?[VE] It seems that you do not share the common often reported issues on digital volume control.
For my opinion digital volume control is better than most analog solutions.
Best would be to position the volume from V200 fully open as here no channel missmatch/crosstalk issues have to be feared.
You can adjust the max. volume by  operation the Pre-Gain adjstments from V200 or the max-gain adjustment from V800.  

3. Using the EM4, which are 124 dB headphones, I am very concerned about noise, is this combo dead quiet?
[VE] It is !! We have several customers using In-Ears who proved that.


4. Ultimately, is it better, since my source will be a computer, to get the V200 with the 24/96 option instead? Are the DAC different? Will you be releasing a 24/192 USB module forthe V200 later?
[VE] Although the USB transceivers inside the HPAs and the DAC are equal, the overall circuitry differs much.
The USB PCB inside the HPAs offer a DAC called PCM 1793, inside V800 there is a resampling unit to completely remove jitter and a better DAC called PCM 1792.
Together with the daughter board for the DAC we will make a 24/192 board for the HPAs

Q&A for Justaudio AHA-120

Here are some questions and answers from Justin of Justaudio regarding their AHA-120 and its use with high sensitivity IEM. 

1. How is the background noise issue? I am very concerned and sensitive to background noise, and was wondering what are the dB levels or noise?

[JA] Measurements are better than specification - in practice with 6dB maximum gain preset and using sensitive IEM (25ohm, 114dB/mW) noise (hiss) is only just slightly (barely) audible at 80-100% volume control position; So dominant noise will be source/mp3 player. 

2. I saw that you do offer a lower gain version for IEM, are there any cost differences?

[JA] Gain jumpers are fitted as standard now, so there is no difference in cost any more if using standard presets (6dB or 12dB).

3. With the pot on the AHA-120, what is the channel tracking like? 

[JA] Very good - with new process it is possible to track within 0.5dB by grading the pots.

4. I read somewhere that you were going to release an IEM specific amp, is there a time frame for that, and will it be a Class A design?

[JA] No plans at present.

iBasso D-zero

Out of impulse today, I laid down some cash for an iBasso D-zero. I really wanted to test out how an amp and USB would perform on the iPhone and iPad, an this seemed to be one of the cheapest solutions. At the same time I got a Fiio line out I connect.

The unit itself seems to be a standard decent construction, nothin terrible or excellent. Good fit and finish. Straight up I plugged her into the iPhone with the line out and started playing music. The phones were the Ety HF5, 105 dB balanced armatures. The gain was set to low. No noise or hiss. That was a great start.

Then I proceeded to play some very familiar tracks through them. The volume control was probably set around 50% or so. First thoughts were better midrange vocals and better extension an control on the bass. That was all I had time for. Now I await the unit to be fully charged and some burning in. But I look forward to more use and integration with the iPad USB. The O2 should also arrive next week, so something to compare.