The new speakers are a side effect of research into electric guitar acoustics. Unsurprisingly the rules for building an electric guitar are similar to how acoustic guitars are built. And of course all of the concepts are the same as any acoustic instrument design principles (a drum or a guitar is basically a piece of wood resonating).

For a guitar to sound good:

1. It can have no paint in the neck pocket and no paint on the neck part that touches wood there. Contact between neck and body must be wood-to-wood.

2. No plastic paint helps. Plastic paint kills wooden resonance. Hence plastic paint must be avoided and wood-based oil varnishes used instead if a guitar (or speaker cabinet) is to sound good. If you have paint in your guitar's neck pocket (and on the neck where it touches the body) - remove it right away.

3. Nuts matter. A brass nut changes neck resonance a lot. This is to be expected as anything placed between the strings and the guitar affects vibration transfer from the neck to guitar. In the case of a nut its material also affects vibration quality (resonance/tone) of the neck itself. Tuners also affect vibration and sustain.

4. Electronics have to be decent, met-poly, silver-mica, poly caps, decent wiring (copper or silver-plated copper), Bourns 1 megaohm pots for volume controls, etc. Makers like Epiphone and Squier often spoil their guitars on purpose with crummy capacitive steel wire and polyester caps to make them sound thick and slow. In the case of speakers Nichicon audio-grade electrolycis or similar capacitors (high-speed/high-slew-rate) must be used for bigger capacitance values (like say 10000 microfarad).

5. Steel on the guitar body gives it a tougher tone, brass knobs give it a tougher classic metal tone, some sort of a glued neck arrangement with a special tenon might give it a woodier/more acoustic tone. Removing the neckplate already makes it sound more acoustic-like, removing bolts and glueing the neck in makes it much woodier and very reminiscent of an acoustic though not quite an acoustic tone.

Violin lacquer is a good idea in terms of paint as long as the lacquer is natural and not plastic. Otherwise natural oil varnishes and dyes are used on both guitars and speakers.

Acoustic guitars benefit a lot from very light wooden bridges (made from the likes of rock maple methinks to avoid getting spent) so depending on how the guitar is supposed to sound removing weight and anything non-wooden off the body is supposed to help an "acoustic" tone.

There's your guide to building a good guitar. Now guess how many of these rules are broken and denied and ignored all the time.

The new Solar speakers are built according to some additional principles:

1. Avoid as many electronics as possible in the current path. That is: No crossover, no preamp/tone controls if possible, and if capacitors or other components such as opamps are used they must be fast (of the fastest slew rate available).

2. Rather than avoiding wooden echoes altogether waveguides are laser-cut wood.

3. Where possible hi-fi fullrange drivers are used to avoid crossover holes

4. Organics everywhere: Paper-cone drivers and wooden boxes are expressive and natural, as they ought to be as musical instruments are made of wood.

5. Instead of damping the drivers and enclosures as much as possible they're allowed to vibrate freely, generating natural wood overtones to music. 6. Wooden legs are used (or screws on the prototypes) to allow natural speaker vibration/consonance.

A fullrange speaker prototype. That sharp angle was a mistake; internally the front bottom joint is more squarish,
with a bass trap in front of the wedge corner.


* Fullrange drivers aren't necessarily "midrangey". They can be very sensitive to damping and enclosure geometry/damping (an enclosure without damping can produce harshness). In a good, dampened enclosure above compliance volume and without excessive driver damping, fullrange drivers can sing. Some drivers (Visaton FRS8M) are very sensitive to back damping and even damping of the panel they're mounted on. As an example, copper tape damping the mounting panel added harshness, Dynamat or Blu-tack damping of the driver kills high-frequency response.

* The other common problem with fullrange drivers is awful capacitors/slow opamps in the amp/preamp. These can kill space/treble. Stick with Nichicon electrolytics, teflon, silver mica, polypropylene and dry tantalium where possible. Opamps must be at least 20 V/microsecond speed.

* Waveguides are a must for rullrange drivers, otherwise treble response will be messed up. Usually stock manufacturer grilles include waveguides.

* Fullrange drivers are generally more efficient than a 2-way crossover setup. This is because in a crossover, almost half the power is burned on tweeter filter's resistors.

* Ported speakers ought to have at least 1/3 the driver diameter port radius.

* Port shape can be anything, even a bunch of small holes, as long as it fulfils the volume/depth requirements of Helmholtz resonance. If anything, the issue is smooth air flow (flanged pipes work better at higher volume).

* Spike (or even screw) legs improve airiness/dimension and provide some extra musicality as long as the cabinet is designed for consonance/resonance (that is, it's a wooden cabinet).

* Avoid sharp angles. They act as standing wave generators, increasing air springiness and firing back into drivers, messing them up in a harsh way. If there are any sharp angles in a speaker, cover them with bass traps.

* Avoid plastic paint. It kills resonance.

* Avoid plastic. In paint, decorations, anything that can interfere with natural wood tone.

* Avoid 4-ohm drivers. They're less stable in high frequencies and tend to overheat amplifiers, especially those not designed for 4-ohm drivers. 8-ohm are stable and have better compatibility.



PCM - pulse-code modulation - is an old sampling format used for waveform signal encoding in digital form.

PCM defines two parameters: Time (frequency) and amplitude (voltage level). Time is encoded by sampling frequency, the horizontal axis of a PCM waveform, in Hz (oscillations/second). Amplitude is encoded by bit depth (bits per sample, BPS).

Two important concepts to understand are thus:

1. Frequency detail is defined by the number of samples available at a given sampling frequency. The higher the sampling frequency, the higher the detail, the more samples per cycle. As sampling frequency rises, there are less and less samples/cycle until degenerating into on/off noise at very high frequencies. The detail limit is f/8. As an example, a sampling frequency of 48000 Hz provides 8 samples/cycle at a signal frequency of less than 6000 Hz (48000/8=6000). 96000 Hz sampling defines well signals of less than 12000 Hz, 192000 Hz sampling has a practical audio bandwidth of below 24000 Hz. Anything below 160000 Hz/24-bit cannot be called "hi-fi" as it does not define at least 20-20000 Hz with enough accuracy.

2. The biggest flaw of PCM is that detail vanishes as bit depth decreases. 16-bit is (unfortunately) a very common BPS value, but it is 16-bit (65536 voltage divisions) only between roughly 0 and -6 dB. Anything below ~-6 dB is 15-bit (32768), anything below ~-12 dB is 14-bit (16384 levels), etc. Things get very coarse very quickly at very quiet levels. Fortunately, there is 32-bit and 24-bit sample detail, unfortunately, modern DACs/ADCs aren't too accurate to realistically fulfil that sort of detail in the real world.



Ed Meitner had mentioned in a Positive Feedback interview (read it, it explains a lot) that in his opinion DSD is a superior choice to PCM because of more natural transients at zero-crossing. PCM zero-crossing has the interesting quality of having no detail at all (0 bits). As it happens though, humans listen to waveform vectors rather than waveforms gradually losing detail as loudness wanes. Once you compare vinyl or tape and DSD to PCM on a modern hi-fi speaker system, this becomes obvious. PCM has a kind of woolly/blurry character to transients, which becomes less noticeable as sampling frequency and bit depth increase, but it is there nevertheless.

These are the basic physics needed to understand PCM sampling. Most digital formats (CD, DVD-Audio, Blu-ray, AAC, MP3, FLAC, ALAC...) are PCM-based. Audio for CDs is mastered with PCM-based tools and stored in PCM wave (.wav or AIFF) format. MP3 files are decoded to PCM for output to DACs in a computer soundcard or portable player, etc. Computer sounds are stored as some sort of PCM waveforms too, such as sounds in games, system sounds, MIDI wavetables, etc. The only digital formats that are not PCM-based are DSD and SACD (which is a form of DSD encoded on an optical disc).

Lossy And Lossless Formats

Lossy formats are a little abomination of the times when bandwidth was limited. The most popular, MP3, is actually an acronym for MPEG (Moving Pictures Expert Group) version 2 Layer III audio compression. Developed by Fraunhofer IIS institute in Germany, MP3 employs a bunch of perceptive encoding techniques to remove audio data deemed "superfluous" in a PCM file. The essential techniques are: Removing harmonics which are considered "excessive" and leaving basic tones (tonality estimation), reducing dynamic range (ATH - absolute threshold of hearing) and removing masking harmonics (muter harmonics occluded by louder harmonics in given ranges), and limiting bandwidth (lowpassing or bandpassing, lower-bitrate MP3 files limited to 16 KHz, 18 KHz...). The result is an encoded waveform which can be about 1/10th the original PCM waveform size. This process is called "lossy compression": Discarding data that in the compression algorithm's consideration is "non-essential" for playback. The result, of course, is a certain thinning of ambience and a dryish/harsher character of the waveform, sometimes with noticeable attack artifacts on chromatic percussion.

The paradox is that sometimes MP3 compression produces more pleasant sound than original CDs. Why does this happen? Simple, CDs are low in resolution and have certain harsh parts which MP3 compression may cut. Real detailed time resolution of a CD is 44100/8=less than 5512.5 Hz. Midrange, in other words. Remember that MP3 compression thins out high frequencies by limiting bandwidth through a lowpass (as low as 16 KHz in a 128-kbps file, or 18 KHz for 160 kbps). These are harsh on a CD: 44100/3=14700 Hz, and there are only two coordinates above that point. MP3 harmonic-cutting algorithm also removes quite a lot of "hard to perceive" high frequencies. Which again, are harsh on a CD.


44100 Hz/16-bit (CD audio equivalent), 12 KHz sine wave. It gets worse as frequency increases; anything beyond ~14700 Hz is technical noise.


96000 Hz/24-bit, 12 KHz sine wave. Notice the larger number of samples allows more freedom to describe treble.

This is very easy to verify by running a lowpass filter on a lossless CD copy (or a 44/16 FLAC download). Classical music works best as strings and winds have a lot of high harmonics. Use NaiveLPF with a VST host like REAPER to experiment with different cuts. You can also use parametric EQ in a player like Winamp or XMPlay to cut very high frequencies. You'll find that 44/16 (and CDs) start sounding more pleasant when the lowpass is around 11 KHz (in other words, 44100/4=11025 Hz). Synthesiser makers have known this for quite a long time, and included analogue lowpass filters to "fatten" PCM sampler synths' sound and allow cutting off the harsher treble harmonics, especially on instruments that don't need them like basses.

There are certain lossy formats which can sound very nice (Opus), but the general good attitude is to stay away from them unless absolutely needed (like reducing a video file size for playing it on a portable device or to reduce download time/size). Other lossy formats are AAC, AC3, Musepack, Opus, WMA (which also has a lossless subformat), Ogg... AAC and AC3 tend to sound tinny with a subtle harshness/dryness. Musepack can be more or less natural, Opus is by far the least distorting lossy format. Ogg can be thinning and cold in sound, sometimes a properly encoded MP3 file can sound better than Ogg. WMA is a proprietary Microsoft format which isn't used much (it was designed to "embrace and extinguish" MP3 but that never worked), it's best to stay away from WMA because it's a Microsoft format. Also as there's no guarantee it'll work on anything (portable players usually just ignore it as an example). In general though all those formats can sound better than lower-bitrate MP3 files as they were all released after MP3 and have improved perceptive coding harmonic-shaving techniques. The downside is a higher CPU/DSP usage, e. g. Ogg requires a faster CPU than MP3, which could be decoded on a faster 80486 CPU (Ogg will need a Pentium in the very least). Memory usage is also higher for newer lossy formats, which is why cheaper devices usually didn't support them (DSP/RAM requirements are higher), combined with developers' apathy towards less popular formats.

A very bad example of an MP3 file (this is 128 kbps, but it sounds more like 64 kbps recoded to 128):



Ogg Q5 encode of the original piece:



Ogg supports 96000 Hz sampling rate by the way.

Original FLAC file:



There are several lossless formats as well, the most common being FLAC and DSD files. Lossless codecs are, quite simply, copying the original digital audio file and compressing it unaltered, without any dirty tricks like washing out harmonics and lowering dynamic range and filtering out high frequencies like lossy compressors do. DSD has been slowly gaining traction as a file format by itself, and many newer DACs and even portable players support it. DSD is not PCM though, it's rather a raw bitstream before quantising (thus it's missing an important distortion stage which is present in PCM). DSD means "Direct Stream Digital" and it is the exact same format developed by Sony used in SACDs (Super-audio-CDs). FLAC means "Free Lossless Audio Codec". It is an open-source, free-to-use codec which in spite of anti-lossless fanatism has been getting popular. Higher bandwidth and larger hard drives and DVD and Blu-ray data discs, as well as large memory cards allow forgetting lossy formats altogether and using FLAC and other similar lossy formats. FLAC is a PCM compressor, at the very basic stage it just doesn't waste bytes describing empty space like PCM wave files do. Other lossless compression formats are Wavpack (having the distinct advantage of supporting 32-bit float compression, unlike FLAC, which can only store integers), ALAC (which looks like Apple's own "divide and conquer" version of FLAC), Monkey's Audio (boasting slightly better compression ratios over FLAC), and WMA lossless, which by now nobody knows exists. All lossless formats except for DSD are PCM compressors, most perform quite alike (don't expect Ogg- or MP3-like feats of 1:10 compression, 0.5 or 0.3 original file size is usually the best they can do, 0.3 being the ratio for a quiet track).

It is good practice to only play and encode to lossless formats, especially now that there are no serious limitations on storage and bandwidth. Lossless formats are also best for processing with any external effects such as EQ (MP3 and AAC files can get toxic with a digital EQ sometimes, other effects like loudness/dynamic boosters can magnify lossy compression defects). Obviously no losses of original data means more ambience and artistic meaning present, at least in higher-res formats like 96/24, 192/24 and DSD.

Psychoacoustics of Frequency Ranges

Everybody knows what frequency ranges there are, such as bass, midrange, treble. However most people outside the music/sound engineering business have a somewhat dim understanding of which exact frequency ranges those are.


Strictly speaking, bass is 120 Hz and down, and it's not directional, anything above that is low midrange,
but ah well. 250-3000 Hz is usually defined as "critical midrange", the range which is clearest to human
hearing. Which is why the "critical range" must be free of any defects in both the mix and the playback
device.


What matters and what most people don't have an idea about is frequency range perception. Bass and midrange are all perceived as "solid" fundamentals and harmonics of an instrument.

Bass:


Midrange:



Treble (5000-10000 Hz):


High Frequencies (10000-20000 Hz):


Fullrange:


Treble is trickier. Treble defines space and presence. What makes a flute sound like a flute is its treble as well as its midrange (not much bass there). Metallic instruments' metallic tone presence is also defined in high midrange and treble.







A string section's presence is defined in treble and even higher frequencies above 10 KHz.


44 KHz/16-bit;


96 KHz/24-bit. Granted, those are two different records, but the presence improves, also sharpness of
definition is better (both dynamics and treble are functions of dynamic/time resolution).


Treble and high frequencies above 10 KHz define ambience, presence, depth, and instrument separation. Very high frequencies are used to give room dimension cues. Treble also conveys emotion and sweetness of instruments.

Differences Between PCM Resolutions


Typical PCM format resolutions are: 44100 Hz/16-bit (CD Audio), 48000 Hz/16-bit (AC97 soundcard standard), 88200 Hz/24-bit (double CDA resolution sometimes used for classical music and live records), 96000 Hz/24-bit (becoming more or less common, this is actually supported by video DVDs as well as DVD-A discs, but seldom implemented in hardware players or even software), 176400 Hz/24-bit (another DVD-A and live/classical music record format, often downloadable as FLAC from online stores), 192000 Hz/24-bit (highest DVD-A resolution), 384000 Hz/24-bit and in theory even higher (fairly uncommon but offered by some online stores and supported by Blu-ray).

There are some quite obvious differences between resolutions which anti-lossless and anti-high-res fanatics will try to deny. Still, they're there. One possible cause for trying to deny them (other than sheer blind, or more precisely, deaf fanatism) is having lo-fi gear or gear that pretends to have some fidelity but doesn't. Accurate treble/HF imaging is a hallmark of hi-fi gear, though nowadays even cheap Bluetooth speakers may have fairly decent treble/HF performance. Most of the physical differences are about treble and high-frequency accuracy/imaging. However as it turns out treble and high frequencies are more important than many people think.

Treble and HF range is essential for instrument separation and space definition. Lynn Olson described his first impression from hearing a CD in the 1980s: "The quiet passages were dead silent... actually, like a switch turned off... but any sensation of space, of stereophonic dimension, and of acoustic presence was totally absent. The start-stop reverberation sounded as flat as a paper Moon and just as fake." This is because CDs cannot define space properly as they run out of samples/cycle in high frequencies. 44100/8=5512.5 Hz - practical detail limit. Being generous might double this range to about 11025 Hz, but anything above that becomes harsh noise that is not very harmonic with the main harmonic (midrange) content.

Because music is harmonic, there are other consequences other than just plain flatness. Expression and life is up there in treble and HF range. Liveliness gets cut/lost when upper harmonics are deformed. Instrument sweetness gets damaged. String sections lose their magic. Cymbals lose their dimension and sparkle. Even bass drums get flattened and lose presence and drums overall lose their "flavour", 3D contour. Overall though, the life of music gets damaged, its meaning and expression. Some bits of meaning are simply not there on CDs, but they are preserved on vinyl and in higher-res PCM and DSD records (easy to prove by listening to a 1970s album in every format).

So in a nutshell, the higher the sampling rate, the finer the grain and the more harmonics are accurately represented. Well not at the 0-crossing, but at louder ranges.

What about bit depth? Bit depth is what defines amplitude detail. Part of the relative charm of, say, 96/24 is that 96 KHz makes it sound deeper and preserves more meaning ("sounds mystical" as an acquainted composer said). But taking the "24" out of 96/24, turning it into 96/16 rather quickly makes it sound cold and hollow, without the same expression. 24-bit allows a lot more voltage divisions than 16-bit, but it still has the same PCM limitation of losing resolution as volume goes down, albeit not as steeply as 16-bit.

Dynamic Accuracy

One of the odd obsessions of audio engineers over the years has been that with frequency response at the expense of dynamics. Accurate frequency playback is all fine and necessary, but what about speed? Poor-quality capacitors in the signal path, as an example, slow dynamics down. The waveform loses liveliness and music becomes more anemic and dull. Dynamic response might be more difficult to measure, but it sure can be heard when playback of a record is compared against a live musical instrument. High-frequency response also relies a lot on quick electronic dynamics. Drums and cymbals play much more lifelike on a fast system rather than a slow one. Everything sounds more lifelike, lively. The problem though is that low-res PCM tends to make things slow and woolly. This has been somewhat worked around by modern equipment manufacturers by using oversampling, e. g. 4x oversampling means an original CD record is played at 176400 Hz rather than 44100. Depending on the resampling algorithm used, oversampling can sound anything from crass and tinny to sharper and more concise than the original CD source. It still won't magically add missing treble definition or missing low loudness definition.

In the '70s and '80s this wasn't quite as noticeable for a bunch of reasons... One, DACs were built with more precision, using R2R designs rather than less accurate and cheap sigma-delta designs fashionable now. So a goodish Kenwood pocket CD player from the '90s, as an example, will still outperform a modern CD player, yet alone an abomination like an MP3 player. There's more depth, dimension and meaning in one of those things. Two, in the 1970s and 1980s designers weren't too aware of the importancy of components, and the components themselves weren't too good (even electrolytic capacitors have improved a lot, not to mention polypropylene, teflon and silver mica). Electrolytic capacitors in the '80s and stuff like carbon-composite resistors all sounded slow, hissy and ponderous. Three, transducer (headphone and speaker driver) technology has improved a lot. You could hardly get the same quality out of a studio speaker driver in the '80s that sold for $200 that you can get out of a mass-produced $15 driver nowadays. Speakers were slow and not too accurate way back in the '80s. Mass-made speaker and headphone driver quality nowadays is stunning compared to even the best designs from 1980s. Four, faster opamps mean modern preamps and headphone amplifiers are quicker and more lifelike. However, slower opamps in the 1980s/1990s also meant digital audio wasn't shown as harsh as it is, as slower opamps attenuate treble and high frequencies where most PCM defects are. Midrange detail is also "bigger" on a slower opamp, so opamp colouring compensated for PCM flaws.

So what does this all mean? In a nutshell, "warmth" and "warm playback" refer to dynamic accuracy as well as accurate frequency playback. Transistor amps also are inferior to valve amps here because the real reason valve amps sound warm isn't just even-harmonic distortion, but latency. Transistor amps are slower in treble and they also tend to distort more in treble than valve amps. Oh sure, the distortion can be more or less suppressed with feedback, but valves will still be quicker and thus more lifelike in treble/high frequencies than transistors. It's very easy to notice by playing electric guitar on a valve amp against a transistor amp: Strings have sparkle and dimension, whereas they get a kind of muddy dullness added on a transistor amp. Even if a valve amp adds its own noticeable colour, it's a less harsh colouring than on a transistor amp. And it can be further reduced with high-quality components.

The important conclusion though is this: Warm playback is defined not only by harmonic accuracy, but also dynamic accuracy. A "warm" device is one that is quick and lively, not just one that can accurately play noises at different frequencies more or less within the same straight line of little deviation. Transistor treble also tends to be slow and thus is perceived as "fake" unconsciously, perhaps one of the reasons why people become less sensitive to treble/high frequencies in records.

Zero-Crossing

First of all, go and listen to a DSD album with some really good hi-fi gear. Then listen to anything PCM on the same gear. Then notice the stilled, lifeless, dull transients of PCM. Playing DSD after PCM is like a fresh summer day in the country after grey concrete urban autumn full of slush. Why is this happening?

Transients.

Us human beings really don't listen to abstract noises as such, we listen to vectors. And our hearing is very sensitive to how a waveform changes, not just its intensity, but which way it's headed from its very birth. Human hearing is extremely sensitive, it resolves down to the movement of an object the size of a hydrogen molecule. In the same way it can resolve a waveform change the size of a hydrogen molecule. Music really is all about those subtle changes as well as power and dynamics and harmonics.

Odd Audio Buffs' Behaviour

One of the odd consequences of the unnaturality of CD sound is an obsession with upgrades, tweaking, and so on, all to improve and get at least a slightly different sound out of mostly flat and fakesy CDs. The problem is that this tends to emphasise some frequency ranges over others, and over time devices start getting around CD/MP3 flaws by making them less obvious: Darkening the frequency response, using slower opamps, even using aggressive comb and lowpass filters to remove harshest frequencies (Apple's approach for its DAPs). On the one hand, this does sort of work, camouflaging the harshest bits out of perception; on the other, better formats cannot play well on a darkened pair of speakers, as an example, as the treble/HF range is mostly cut off, so presence, expression and depth are gone.

This situation is obviously a vicious advantage for manufacturers, as they can keep pushing different kinds of sound and digital processing as new fashions. The main problem here is a vicious "democratic" approach to judging format acceptance by the likes of Sony and Philips: "If regular listeners can't make out differences between high-res and low-res PCM, let's leave the formats expensive and hard to source for those willing to shell kilobucks out". Never mind the utterly idiotic situation when a cleaner woman at a cafe gets moved by 192/24 classical records. How they manage that sort of a prejudice is a mystery, perhaps they just want to kill music with a dead format like CD audio and derivatives. Needless to say, everyone ought to make every effort to spread high-res formats in spite of this vicious anti-musical attitude.

Myths And Misconceptions

1. "Higher-res PCM formats' very high frequency resolution matters a lot". E. g. 192 KHz PCM in theory defines slightly less than 96 KHz signal bandwidth. Quite the opposite is the case, main hearing of most people deteriorates with age. Older people cannot hear much better than 16 KHz, even. Now there were experiments which have proven that humans are also affected by ultrasound (e. g. a forest noise's ultrasonic components affect brain state and give that soothing calm feeling), but in reality a higher-res digital format's frequency resolution matters much less than the finer sample grid. Worse, many amplifiers (and DACs?) are unsettled by very high-frequency harmonics. So it's only prudent to lowpass even 192/24 audio at a certain frequency, say, 30 KHz, to remove harsh harmonics on the analogue playback side. So to restate, it's the finer coordinate grid that matters (more freedom/definition for the waveform) and not high-frequency harmonics as such.

2. "Digital formats and signal transition are flawless". This is obvious nonsense. Digital encoding and decoding with a lot of processing (such as quantising) in-between are another kind of distortion that the originally analogue (in most cases) signal has to go through. A microphone's output is an analogue signal. Quantising it to a finite number of digital samples will inevitably make it coarser than the infinite-coordinate original current shape. Besides which, there's a bunch of ways in which even digital signals can be distorted, ironically because of an interruption in analogue modulation of digital transmission.

3. "Newer is better". Often it isn't. As an example, LDHC, Bluetooth's lossless PCM transmission protocol, only allows 96/24 PCM transmission maximum. Yet Bluetooth headphones and speakers have been getting popular in spite of their technical inferiority. Besides which, one would rather have a nicer proven DAC (like AK4556) handling decoding rather than a colder/harsher design in a Bluetooth speaker.

4. "CD audio has a wider bandwidth than analogue tape". By now it should be obvious it doesn't. Said "22050 Hz" signal bandwidth is a scam as it should be obvious from what was written above. Sometimes fanatism becomes ridiculous, a fellow in a tape deck group was once complaining about how his tape deck could not go "beyond 23 KHz" unlike a CD. Now strictly speaking type I cassette tape would really max out around 16-18 KHz definition, but those were real moving signal KHz, not CDs' fake 1 or 1.5 samples/cycle at the same frequencies. That's more of a "noise bandwidth" rather than practical musical record bandwidth.

5. "Digital distortion doesn't exist". Oh yes it does, plenty of it. The big difference is that most analogue distortion is positive, it adds up to the original signal. Digital distortion is negative, it deforms the original signal by not describing it accurately. Original signal is audible under analogue distortion, but you don't know what the original signal was under digital distortion unless you compare it with the original master. Good luck if it's open reel or DSD or high-res PCM; otherwise treble and low-volume details are lost forever. The best way to compare different formats is to listen to real instruments, record them with goodish microphones and then listen to the records.

6. "The same impression is conveyed no matter the format". That really is the problem, it isn't. Lower-resolution formats kill the magic. Meaning and artistic expression get lost with details. There's something in the treble that vanishes in CDs and the like. MP3 files are just plain dead, lifeless, music-less most of the time. The problem is that in many cases it's the electronics that kill definition too; consumer audio devices are just too coarse all too often.

7. And this leads to myth 7. "You need hi-fi gear to even hear the differences in high-res formats". Quite the opposite is true. CDs and even MP3/AAC files sound better on hi-fi speakers/headphones. 96/24 PCM already plays more lively on cheapest headphones, noticeably better than CDs.

8. "But there is no noise as with vinyl!". Part of the problem with this attitude is that the goal of a record is to transmit as much of the original performance's magic, art as possible. So the real difference between formats is how much of the original performance is transmitted, rather than any noise. Which is what many people don't listen for, even though they ought to.
A note on amp and preamp configurations: Passive components, resistors and capacitors, have a major influence on sound. In straight terms, to sound musical, an amplifier has to transmit current as swiftly as possible. Poor-quality capacitors and resistors will slow current down. Big steel resistors can make things awfully dull. Capacitors have two distortion parameters which matter: ESR, equivalent series resistance, which is basically a resistance that can turn a capacitor into an auto-lowpass filter, and DA, dielectric absorption, a memory effect that can blur sound waves with leftover laggy current. Both ought to be as low as possible. Polyester capacitors tend to have a high DA value and so are mostly junk. Cheaper electrolytics can have a fairly high ESR, lowpassing around 8-10 KHz or even lower, or slowing high frequencies down. Multilayer ceramic capacitors can sound nasty too.
An amplifier or preamplifier will therefore sound only as good as its worst capacitors (or resistors). Metal film resistors should be used, though again, steel by itself can lag dynamics something awful (steel is a much worse conductor than copper or silver), so larger resistors ought to be avoided where possible, and small SMD resistors are always better than long-tail radial ones. Audio-grade electrolytic capacitors should be used (Nichicon preferred) where electrolytics cannot be avoided, and silver mica, polypropylene and teflon or even dry tantalium where affordable.

Amplifier ICs

Texas Instruments TPA3316D-2



A class D amplifier, this one is tricky. It can sound passably good, but this depends a lot on output (and perhaps input?) filtering. Filters have to be optimised for a particular speaker impedance rating. An amp board configured for 4-ohm speakers will sound fogged with 8-ohm speakers, this also depends on a particular driver's impedance curve, impedance rising steeply in treble usually makes speakers fatiguing. Fatigue is another issue with this amp chip, even 4-ohm speakers can get stressing/tiring after a while. It's a subtle effect, but usually the more tired the listener is, the worse it gets.

Yamaha YDA-138E




Class D. Highly sensitive to interference, very sweet sound for a class D amp. Sparkling treble, especially after the TPA3116, which is darkish. It must be powered either by a very filtered power supply or a battery (12V to 14V DC). Most cheap boards on sale have filters optimised for 4-ohm apparently, but it also sounds good with 8. Yamaha recommends 1-microfarad bypass capacitors on input, which can also boost bass in addition to cleaning line noise. Apparently those capacitors also affect character; dry tantalium added a lot of bass as an example, but also sounded a bit harsh, giving that class D fatigue similar to TPA3116. Silver mica or polypropylene might be better? The sneaky suspicion is that shielding speaker wire might reduce interference noise which it absolutely loves to inject in bass/low midrange, even in a Bluetooth amp with no line input.

National (TI) LM1876



Very sweet-sounding chip, approaching class D transient/drum detail without the fatigue. Possibly one of the best mid-power class AB chips. Its only flaw is it gets a bit anemic (not enough bass/low midrange) without enough power. Very nice treble. Sounds absolutely great with 1-microfarad silver mica bypass capacitors on input and Nichicon electrolytics.

Preamp/Headphone Amp Opamps

Radio Company of Japan JRC4558



As the name states, and as one might suspect, this chip was created for cheap transistor radios. It sure sounds the part; forward midrange/bass, next to no treble extension, slow slew rate. Something curious about it is that it does image treble extension with DSD (there's space definition) but not as much with PCM output, not just CDs but also 96/24 and even 192/24 PCM. Talk about PCM fake reverb.

Guitar effect tweakers/makers call it a "warm" opamp. It might help electric guitars which have a practical midrange output (your average guitar amp cuts off around 5000-6000 Hz), and it does exaggerate bass/midrange detail, but at the cost of speed and treble. Great in a distortion pedal for sure.

Dynamics aren't quite as good on this one, and again, it tends to sound like an old radio when playing anything hi-fi (like DSD). Curiously, vinyl doesn't sound as bad through this chip in a preamp though. A bit darkish but passable.

Asian manufacturers love sticking it into amp boards, like the TA2020 class D little amps. There it tends to sound a bit cold/anemic. They also stuff it into anything from TDA7265 and TDA7294 to triode valve amps. Which is daft as a good valve amp's main value is quick dynamics and treble extension, which the JRC4558 happily mangles.

One reason for its popularity is that it "dresses up" low-res PCM-based formats like CD audio and MP3/AAC files, hiding the ugliness of slow transients and lack of treble detail by slowing it all down into a crawl (this is a 1.6-2V/usec chip). The drawback is of course that this backfires on vinyl and DSD and tape, which have much better treble/dynamics and real space separation/imaging.

So, in summary, warmish/slow (yet punchy) midrange and bass, slow and poor treble/space imaging. It might be good on bass/midrange drivers, and be combined with a faster opamp (OPA2134?) in a tweeter preamp/amp in a biamplified design. It's a very '60s-coloured sound though.

Burr-Brown (Texas Instruments) OPA2134



This is one of the nicest opamps out there, supposedly slightly worse than the OPA2132. It's not the cheapest design, but thanks to its faster slew rate (20 V/usec) it is quite suitable for music. In a good setup with fast, high-quality capacitors and metal film resistors it sounds very lively, almost 3D in imaging, while also making drums big and having a kind of a mellow ambience presence. Cymbals might be slightly sluggish compared to no preamp. Great in a headphone amp, especially with older 600-ohm headphones like AKG K-140 which sound very realistic even in a simple CMoy setup (just give them a good shielded cable). This is an obviously coloured amp, but in a pleasant way, with even harmonics giving it a cosy reaction/ambience, its slew rate still giving it enough treble/dynamics/space definition.

National (Texas Instruments) LM4562



Similar to OPA2134, a slightly coloured, yellowy-warm sound. 20V/microsecond. Less candy-coloured than the OPA2134, more on the slower side as well. Some people call this a "neutral" opamp, well it's still coloured, only less so than slower opamps. It just isn't a "fun-ride" opamp like OPA2134. Cymbals can be less realistic, lacking sparkle compared to unamplified or a faster opamp, somewhat more towards "copper/brass" fundamental tone rather than "trebley sparkle". Overall its character is "stable", It can add some 3D definition that might be lacking without it or on a more neutral opamp to records that lack it (such as those with '80s samplers). Drums are a bit on the slower side, but weighty and defined. Bass can be a bit heavyish. Everything gets somewhat more punchy and strings and synthstrings have a more palpable, textured presence as a plus. Guitars are fine, a bit slow and lacking excitement, but it transmits the roar/distortion texture. A bit mellow/thick in the upper midrange, a bit heavy/stiff in bass. In author's opinion, LM6172 might be a better choice for a preamp because of its speed and transparency. LM4562 might be a good replacement for the NE5532, it's still vaguely neutral/analytical and a bit slowish/static; it is "contoured" in a word. Some say it's "vaguely cold"; it isn't, it's more like "slightly static" at times, especially in bass.

Signetics (TI) NE5532



It's that '80s CD player sound. Glassy, nothing really stands out. No impact. Slow, but not as slow as to kill space altogether like JRC4558 does. "Dull" is a good description, some people call it "neutral". It used to be stuffed a lot in SSL consoles and Adam Audio monitor speakers. Also some TV sets and VCRs. It can be harsh with certain DACs (48 KHz and less sampling rates; it's mostly fine with 96 KHz and up) where it creates an ugly odd-harmonic treble fog. Too boring/slow for fast modern speaker drivers. IMHO it's one of the first candidates for replacement. "Mediocre" is another good description. It might be useful in monitor speakers to create a lively mix (rule of the opposites), but it's not good enough for any device that is meant to play lively/realistic. Drums tend towards "splats" and cymbals lack sparkle and metallicity, sounding more like tin midrange (or misaligned-head tape deck). Everything like sizzling analogue synth strings/pads tends to be mellowed down with a dullish temper. It can hide some flaws/low-res PCM harshness in the same way JRC4558 does, by not being detailed. Funny bit is, JRC4558 sounds more interesting thanks to its thick, contrasting (if slowish) midrange/bass. Replace with LM6172 for transparency or some of the OPA series for colour. Don't bother using it in a new design unless it's dull on purpose, like mixing/mastering gear.
Note: These were originally copied by ETTiNGRiNDER from the Raven Games website. The interviews appeared at http://www.ravengames.com/heretic/insider.php and http://www.ravengames.com/heretic/heretic-shadow/insider.php. The text has been slightly edited to fix spelling/typing mistakes.

Heretic insider info: Behind-the-scenes interview with Michael Raymond-Judy

Heretic might be an old game, but some of us still love it. For this reason, Ramborc made a behind-the-scenes interview with level designer Michael Raymond-Judy of Raven Software. Note that before the interview it wasn't known that the maps for Heretic have been made by several people (Mike is listed as the only level designer in the credits of Heretic), thus some of the questions seem to be contradictory.

Hi Michael. You were Level Designer on Heretic. Exactly what tasks did this title involve?

Well at that time it was pretty much just making maps and placing enemies and goodies. I did help a little with story background and defining locations, enemies and weapons/items, but since that was really the first project I worked on from start to finish the bulk of my work was map-making. I also got to "help out" with some other people's maps doing stuff like aligning textures (the words "firstcol" and "firstrow" still give me shudders) and making sure there was a balance of ammo to enemies.

Heretic was quite a rough change compared to previous fantasy games from Raven. It gave up on the "thinker" RPG aspects and delivered an instant action shooter with a couple of extras like an inventory system with artifacts. What were the reasons behind this decision?

I think the biggest reason was the more direct involvement of id Software. They had some pretty strong ideas about what they liked in games, and since they were the ones providing the technology we listened very closely to what they said. Also we had been "beta-testing" Doom for a while (which is to say we spent a lot of hours running around slaughtering each other in deathmatch) and we saw how much fun that kind of game was. The inventory was probably the main holdover from our RPG mentality, and I think it did add a new dimension to the FPS style of play.

Where did the ideas for the world, creatures and background story come from?

Some of it came from us, some from id (Sandy Peterson I think wrote the little bit on the back of the poster/guide) and some, sadly, was made up by the people who later wrote the manual/hint book. I say sadly because they pretty much made things up as they wanted without asking anyone here, and a lot of what they made up just didn't fit with the "reality" we had created behind the game. Like the second Highlander movie, I just try to pretend it never happened...

I can't help but notice that Heretic made a lot of things nearly the same way DOOM did. No, I'm not saying it's bad... rather the opposite. I myself think that part of Heretic' success was that it capitalized on the factors that made DOOM such a hit. Were these similarities done consciously?

See above :)

(I refer to things like the kind of weapons you get, the kind of monsters, the structure of each episode like the feeling of the start level, that the last map of an episode (ExM8) is rather small, but the last but one (ExM7) is a VERY extensive one, the kind of bosses in the end maps, etc.)

Ditto :) With the addendum that some of this was inherent in the code we got, and since we didn't have time to change some of it (how many maps and what they are called, stuff like that) we were stuck with it. We DID increase the maxvisplanes to about 4x its original, to accommodate the *ahem* "ambitious" layouts of some of our maps (when the boss builds something, you try to make it work).

What were the main advantages and disadvantages of the DOOM engine? Were there things you planned but had to scrap because of engine limitations?

At the time the Doom engine was so far beyond anything else out there we really didn't hit too many limits (other than the aforementioned maxvisplanes). The worst thing I ever had to do was take one map (not mine) and put a big honking wall right in the middle of it so you could actually run it. I think it was E1M4 (the big city "square" with lots of buildings and castle walls outside). [It's actually E1M5, "The Citadel" — editor's note.] Nice map. Too complex. As for advantages, I have said and will continue to say that working with a known technology and existing tools (no matter what quirks they have) is less sweat than trying to develop engine, tools and game simultaneously. A known limitation is always better than a surprise, since you can at least plan around it.

Through what changes went the game through in the course of development?

Pretty much the biggest change was the name. I think it was called variously "Mage" and "Orb" and a few other things (I think "Vorpal" was the one that got the worst reaction from people here — 'it sounds like a whale fart' someone said). Other than that we just had to re-make or re-colour some items and monsters, and add one weapon (the Firemace, or "lobby ball" as we called it); the rest of the game stayed pretty much intact from start to finish.

How did you create the maps?

Very quickly. I think on average I could crank out a first run at a map in about 3 days, including textures and initial placements. Then it took a few more days (sometimes as much as a week) to tweak textures and items to make it balanced and pretty. Then the boss hated it so we made a new one :)

The editor (Doom-Ed) was a breeze, other than the fact it had no side view, and aligning textures was a pain. It ran on our "power machine" of the time, a 486SX running on Nextstep. Zoom.

What did it feel like to create all the maps of a whole game alone?

I never did that, but I did touch every map that went out. I think I ended up making a bit more than half the maps for the original 3 episodes, and thankfully Eric Biessman got hired and he cranked out a bazillion maps for the SOTSR add-on. But I can say that FIXING all the maps for an entire game, especially ones done in a hurry by someone else, is, well, "not fun". Except of course that I got to add little secret rooms and traps to their maps and move weapons for deathmatch, so when we played the "builder's edge" suddenly shifted to me :) (or as someone one said, "Now I am the master!").

The original Heretic had a secret map at E4M1. Who did create this map, what was the reason for it, why hasn't the map been finished, and what name would you've given it if you'd have had to?

It was supposed to be a deathmatch-only map, I think it was patterned off a Doom DM map (built by American McGee I think). There were so many versions of Heretic put out (freeware, shareware, retail, add-on) that I think it got lost in there somewhere. I don't think I ever thought of a name for it.

If you could go back and redo your work on the project, would there be something you'd do different?

I'd push to have a lot more maps, and make the retail version include about 8 episodes. We dumped what I thought were some good maps (especially DM maps) because we didn't have space on the floppies... Also I'd spend more time on the bosses. Like several of our games I felt more had gone into the sub-bosses (I loved the Maulotaurs!) than the main boss, because the main boss always gets done last when there's no time to do it right.

Thank you for your time.

No thanks necessary, just send scotch ;)

Heretic Shadow of the Serpent Riders insider info: Behind-the-scenes interview with Michael Raymond-Judy

Heretic might be an old game, but some of us still love it. For this reason, Ramborc made a behind-the-scenes interview with level designer Michael Raymond-Judy of Raven Software. Note that before the interview it wasn't known that the maps for Heretic have been made by several people (Mike is listed as the only level designer in the credits of Heretic), thus some of the questions seem to be contradictory.

Hi Michael. You were Level Designer on Shadow of the Serpent Riders. Were your tasks any different from those on Heretic?

Not really. Although there were more people helping (good thing) which meant sharing the map load eased things a bit, it also meant more watching what everyone was doing and making sure it was up to standard and worked together well.

What was the decision behind creating 2 new episodes for Heretic and releasing them? I see a strong similarity to the re-release of DOOM together with a new episode as Ultimate DOOM. Was it a conscious decision to follow id Games' example?

Actually it was more that we needed to fill a "project gap" until we got Hexen contracts worked out, and we had a good demand from the public for more material. I also felt personally that SOTSR was what Heretic originally SHOULD have been — more episodes, more content, and a wider variety of areas. In some ways it would have been nice to re-order the maps and move items so you could play the episodes with D'Sparil at the end, but that's not how it worked out.

How did you work together with Eric, the other level designer? Did you create maps together or did you make your maps separately and then put it together to form the 2 episodes?

We did some of both. We also traded maps back and forth to get a more "balanced" feel, so I would work on areas he had problems with and vice versa. It's a way of making maps that seems to have gone by the wayside (too little time now I think) but personally I felt it made all the maps feel more polished. Having more than one perspective is always good, and when you built something yourself it's hard to step back and say "it doesn't work". Letting someone else rip it apart and rebuild it may hurt your pride, but it makes better gameplay.

Shadow of the Serpent Riders has 2 secret maps at E6M1 and E6M2. Who did create these maps? What was the inspiration for them? What name would you give these maps if you'd have to?

E6M1 was a map based (very loosely) on the old Raven office. So maybe "Raven's Lair" would work for that one :) The other one was built by Brian Raffel, it's kind of a mix of ruined temple and mine, so I have no idea...

Thank you for your time.

Just send more scotch ;)