Monthly Archives: March 2019

Planar Speakers

More specifically, why I like planar magnetic speakers (and headphones!).

Sound quality: this one is subjective, yet important. When set up properly, planars sound more natural, open, and transparent than conventional speakers. They’re perfect for acoustic music across all genres from small to large ensemble classical, jazz, vocals, etc. Solo piano, vocals and chamber music are particularly good on planars. Their midrange is uncolored, having incredibly high resolution, yet without the artificial detail of boosted upper mids/treble, and without adding the glare or edginess of conventional dynamic drivers — unless that edginess is in the recording itself! With the 3.6/R I frequently hear subtle musical details or tone/balance shifts that I never hear even on the best headphones. Music is mostly midrange, and that is what planars do best. And the treble is simply astounding. No speaker on Earth matches the high frequency extension and linearity of those huge ribbon tweeters. The transition from the mid panel to treble ribbon is seamless, preserving the timbre and harmonic structure of acoustic instruments and voices. And that bass… clean, tight, with a seamless linear transition from the mids.

Low distortion: Measuring total distortion in Room EQ Wizard, my  Magnepan 3.6/R measure about -60 dB (0.1%) in the treble, -50 dB (0.3%) in the midrange, and -40 dB (1%) in the bass (at 60 Hz). That’s lower than most conventional speakers, even lower than most headphones. And it is an uncorrected figure, including the distortion in the microphones, amplifier, and DAC; the actual distortion from the speakers alone is even lower. The Audeze LCD-2 headphones (planar magnetic) measure < 1% total distortion throughout the entire frequency spectrum, even to sub-bass frequencies. No conventional headphone matches that, not even the Sennheiser HD-800.

Why is planar distortion so low? I can think of 2 reasons. First, each Mag 3.6 panel spans the area of about six 12″ woofers, and its ribbon tweeter is 5′ long. Such physically large drivers take only very small movement/excursion to produce a given sound level. And the distortion that a driver produces is related to its excursion. Second, the drivers don’t have as strong Q resonances as conventional drivers do, both mechanical and electrical.

Linear phase: The 3.6/R have a relatively flat impedance curve: 4.2 ohms in the bass, to 3.3 ohms in the treble. They don’t have the big impedance vs. frequency swings that conventional speakers have. This promotes linear phase and flat group delay.  The 3.6/R measure group delay of a flat zero through most of the frequency range, and only exceeds 10ms in the bass (below 80 Hz).

Easy load: Because planars have relatively flat impedance vs. frequency, they are primarily resistive loads that are easy for amplifiers to drive, despite their lowish impedance.

Drawbacks

Planars are dipoles, so they radiate equal energy front and rear, and the rear energy has inverted phase. This makes them more sensitive to room setup than conventional speakers. This can be a blessing or a curse, depending on your situation.

Planars tend to be inefficient, so they require more power for the same listening level. However, their dispersion is line-source (rather than a point-source), so the volume does not drop with distance as quickly as with conventional speakers.

Planars have limited maximum loudness. In a medium-large listening room, the bass distortion of my 3.6/R begins to rise at 95-100 dB SPL (and requires 400+ watts per speaker to attain). This is plenty loud enough for me, but it’s not for those who listen at ear-shattering levels.

Planars are difficult to measure because near-field, you can’t “hear” all the drivers from a single microphone position. And far-field, what you measure is as much the room as it is the speakers.

Planar drivers are side by side (the panel and the ribbon tweeter). They can’t be aligned vertically like conventional speakers, so the midrange to treble timing and impulse response depends on the angle between the speakers & listener. More specifically, the speakers should be angled so the panels are about 2″ closer to the listener than the ribbon tweeters.

Planars usually require a big room, and sound best when placed well into the room away from the walls. This leads to a low wife-approval-factor, and requires a dedicated audio room.

While planars have taut, low distortion bass, they usually don’t reproduce the lowest octave. The larger ones, like the 3.6/R, are good down to about 30 Hz, and 25 Hz is clearly audible though attenuated, which is fine for most music. But if you want that room-shaking 20 Hz rumble for movies with explosions and such, you’ll need a subwoofer.

Meier Audio “FF” Frequency Adaptive Feedback

Meier Audio has a feature in their amps called “FF” or Frequency Adaptive Feedback. Jan Meier describes it here. His article is detailed yet long. I wrote this article to complement it to help in understanding.

Musical Hearing

When it comes to human perception of sound and music, all frequencies are not created equal. The ear is most sensitive to frequencies from around 600 to 3000 Hz. And, most music (at least voices and acoustic music) is concentrated in this range.

Consequently, this is the most critical range for reducing distortion. You probably cannot hear 1% (-40 dB) distortion at 60 Hz, but you can hear it at 2000 Hz.

Analogy: Dolby B and RIAA equalization

Readers with a few grey hairs remember cassette tapes and Dolby B noise reduction from the 1970s and 80s. Dolby B was brilliant in its simplicity. Tape hiss has a wide frequency spectrum but it’s most noticeable in the treble (this is where our hearing is most sensitive). If you cut the treble during playback, it reduces hiss but it also dulls the music. So when recording, boost the treble. Then during playback, cut the treble by the same amount you boosted it. You get the same hiss reduction without any reduction in treble, because you’re only cutting exactly what you boosted earlier. The music has flat frequency response and sounds cleaner with higher S/N ratio.

The RIAA curve does the same thing for LPs. The pre-emphasis equalization curve cuts the bass relative to the treble before cutting the record groove. This reduces the groove and needle excursion needed to handle low frequencies, reducing distortion and noise. On playback, the phono head amp applies the opposite de-emphasis equalization curve, restoring flat frequency response.

The main drawback to this is that boosting the treble when recording limits the dynamic range. You can only boost it so far, before it reaches peak levels and overloads. Boosting the treble may require you to reduce the overall recording level. Alternately, reducing the bass lowers the SNR of the bass. Yet it improves the SNR of the treble, and this is a desirable tradeoff since that is where our hearing is much more sensitive to it.

Amplifier Feedback

Solid state amplifiers have a negative feedback loop that reduces distortion, increases bandwidth, and increases stability. Contrary to what we may read in some audiophile circles, negative feedback is A GOOD THING.

What exactly is negative feedback? An opamp’s native or open loop response, gain-bandwidth curve or transfer function, is not linear in both frequency and amplitude. So a portion of its output signal is inverted and fed back into the input, which offsets these non-linearities.

Furthermore, an opamp’s open loop response drops with frequency, around 20 dB per decade or 6 dB per octave. This means negative feedback has much stronger low frequencies than high frequencies. We can quantify this. Human hearing from roughly 20 Hz to 20 kHz spans a frequency range of 1000:1, or about 3 decades. So negative feedback is roughly 60 dB stronger at 20 Hz, than at 20 kHz.

More on negative feedback here.

This means most of the benefits of negative feedback are focused in the low frequencies. Higher frequencies have progressively less negative feedback. But perceptually, we want the opposite! Distortion & noise are much easier to hear in the high frequencies. So applying a pre-emphasis curve to the signal, similar to what RIAA does for vinyl, can be beneficial in the gain-feedback loop.

Frequency, Energy and Amplitude

Most of the amplitude in a musical signal is in the low frequencies. The midrange and treble, where our hearing is most sensitive, is just a smaller ripple riding on the much bigger bass wave. Reducing the amount of bass shrinks the entire signal, without any loss of amplitude or resolution in the midrange and treble. This keeps the signal away from the near-full-scale amplitude swings where devices get less linear.

This is particularly true of DACs – they get less linear for near-full-scale signals. Reducing the amount of bass before D to A conversion, then boosting it back afterward, can reduce distortion by keeping the DAC operating in its most linear region.

Frequency Adaptive Feedback

Combine these 4 ideas and you have Meier Audio’s FF. Start with the musical signal.

  • Step 1: pre-emphasis: boost the critical frequency range (midrange-treble)
    • Alternately, attenuate frequencies outside this range. This can be a better approach since attenuation means no chance of clipping.
    • This is the first thing you do when the signal enters the amp.
  • Step 2: pass this emphasized signal through the amp’s gain-feedback loop
    • Or through the DAC for D to A conversion
    • This weights negative feedback effects toward the critical frequency range
    • This reduces the signal from near full scale to the DAC’s more linear region
  • Step 3: de-emphasis: attenuate the critical frequency range
    • Do the reverse of what you did in step 1.
    • This is the last thing you do before the signal leaves the amp.

In summary, FF has 2 potential benefits:

  • Compensate for negative feedback’s bass-heavy content, giving relatively more correction at midrange/treble frequencies
  • Reduce signal level to stay below peak levels having higher distortion, without reducing midrange/treble resolution

FF can be particularly effective for modern recordings which use heavy dynamic range compression with peak levels near full scale, or even have intersample overs or clipping.

Incidentally, the Redbook CD specification has something called “emphasis”, which is similar to FF. It boosts high frequencies (from 1 khz to 20 kHz). CD players are expected to attenuate those frequencies on playback. This is akin to Dolby B for digital audio.

Counterarguments

Here we’ll play some devil’s advocate.

If distortion is already below audibility, then FF is a solution looking for a problem – what is the point? In fact, the cure could be worse than the disease! FF requires filters on the input and output to shape the frequency response. These filters cause their own distortions (such as phase shift from analog filters or minimum phase digital filters). The overall effect is a trade-off between the benefits of FF and the drawbacks of having this extra signal processing.

Most opamps have far more gain than we need, so we must use a lot of negative feedback. So much, that the bandwidth is several times wider than audio, 100 kHz or more. Thus, even high frequencies have enough negative feedback to reduce distortion below audible levels, even if they have less feedback than low frequencies.

FF actually increases distortion outside the critical frequency range! With FF you will have higher distortion at lower frequencies (because FF attenuates them in the feedback loop). But you’ll have lower distortion in the midrange and treble. FF shapes distortion to match the sensitivity of our hearing: less distortion where our hearing is most sensitive, at the cost of higher distortion at low frequencies where we can’t hear it.