The LM-1 vs. Stock

Unfortunately it seems my designs are so popular that in the US the LM-1 and its sibling the LM-1C have caused Madisound and Parts-Express to run out of stock. Fear not, Parts-Express expects more in February of 2017.

If you absolutely cannot wait, the Peerless 830860 will substitute. Same motor with a polypropelyne cone. Probably won't be as transparent, but I've not heard or measured it. Specs say it's going to be very close.

Lepai 2020A Micro Amplifier

I purchased a Lepai 2020A 20 watt micro amplifier to power the LM-1 speakers I recently put on my desktop.I used the micro plug for the PC sound and the RCA inputs are being fed from my Logitech Squeezebox Touch.

The Lepai 2020A is part of a family of amps that visually look identically but are based on different chipsets or have higher power ratings.  The "A" variant is based on a Yamaha chipset. Yamaha has been making professional amplifiers for decades, including very high efficiency models for studio and touring use.  The 2020A uses a 12V/3A supply.

Based on a variety of reading material, I doubt the 2020A does 20 watts/channel. It's probably closer to 7 when you limit power rating to 1% THD. Still, sounds very good.

For the price, it's really kind of ridiculously good.

The next model up, the 2020Ti takes up to a 24V/3A supply and gets closer to 14w/channel at 1%. Sort of.

The sound is smooth, very quiet, and not fatiguing at all. It has plenty of power for watching Hulu while in bed or playing games, as well as listening to Jazz FM 91 via the Internet.

The tone controls are subtle! They may not do it for you if you are looking for big changes, but if you like to gently adjust bass and treble they'll be great for you.

Some users report even better performance with beefier power supplies which I did not try. In my case small and hidden is key! Plus I never really felt the amp was too small. One thing though, the LM-1 speakers are VERY easy to drive. Your success with speakers that have lower impedance may not be as good.

The Lepai 2020A does not have any input switching. Both the RCA and mini jack inputs are live at all times. Sadly they are mixed through a resistor, not buffered. and then mixed together. This means that if two devices are operating, you'll get half the volume of either. Turn one off and the volume doubles.

If you don't intend to head bang, but are looking instead for a solid mini-amp to put under your desk, or in a boat or RV, I highly recommend the Lepai 2020A. 

The amp runs ice cold at all times. The one negative thing is this desire for Asian manufacturers to put blue LED's on every damn thing. In this case it's around the volume control. It's big, bright and blue, so you can forget about sleeping with it on nearby.

Stereophile Hatchets the Crystal Cable Arabesque Minissimo Diamond

This is kind of a follow up on my original article Stereophile - The Data Doesn't Lie and, to be honest, this is a tempest in a teacup as well. It's just one sentence that lit a fire under me.

First, I've never heard the Arabesque Minissimo, nor can I really morally justify the purchase of small home speakers that cost $20,000, whether from Crystal or Magico.

Even then I am nothing if not a hero for truth and justice so it was inevitable that the latest online review from Stereophile would completely piss me off. For the most part, it's a too crappy and short for a pair of speakers with such pedigree.

Stereophile Listening Tests 

The listening section is only one page, and very little devoted to actual music listening. Also, John Atkinson seems to write that he relies on test tones and his ears to tell "tonality" instead of oh, I don't know, music? I mean, your ears are for music, not test tones. If you have the gear, you don't need to be using test tones and aural guessing.

The biggest problem is that by JA's own admission, the speakers could not be placed optimally in his room. Instead of realizing there was significant "operator error" JA goes on to blame the speaker's and claims "treble tailoring" will only play nice with some music.

So what happened? Speaker/reviewer mismatch.

To have the maximum possible sensitivity and simplest crossover it seems that the Minissimo foregoes "baffle-step compensation" or BSC, and requires close wall placement. I have highlighted this region under the line in the chart at the left. This rising bass effect happens as the woofer's output goes from omni-directional to half space, and is related to the woofer and the baffle width. Notice it does not show up in his near field graphs...that's a whole other story of mess too long for this post. 

You might look at that and say "what a crappy speaker!" but you'd be wrong.  There's nothing wrong with designing a speaker for a specific environment, and we need more speakers that don't try to dominate your living room. This kind of design approach can yield exceptional results. Unlike almost all "high end" speakers, the Minissimo's were never designed to be a "one size fits all" model. They are ideal for a salon where discretely beautiful speakers could play wonderfully while leaving space for everything else. Listening to the Minissimo's in the middle of the room will sound like a mosquito orchestra, which seems to be just what happened.

JA rigidly reviewed them as middle-of-the-room stand mounts and blames the speaker when he should be blaming his closed mind and lack of understanding. With these measurements in hand, JA should have stopped or found a better place to listen to the Minissimos, or held of on publishing the review. Instead he attributed the weak response to the "tailored treble." His excuse that his furniture did not allow him to place the speakers correctly is the height of professional whinery. Either move them to the right place, or let some one who can and wants to them properly so. Don't compromise by writing a half-ass review with the speakers in a poor location.

The Bottom Line

The Crystal Cable Minissimo designers took truly refined components with a minimalist design aesthetic and made a high quality, beautiful looking pair that will stay out of your living space. Comparing the Minissimo to the slightly less expensive, but equally insensitive, Magico S1 Mk. II, the S1 requires much more floor space to sound it's best. To me, a small speaker that actually needs so much space is kind of a weird situation. On the other hand, the Minissimo is small (for a speaker with a 6" woofer) and perfectly happy to get cozy to the art on the wall.

With this in mind, if you are a music lover who wants top tier, beautiful speakers made from rare materials that don't take over your room I strongly encourage you to listen to them for yourself and pay no attention to the critic behind the curtain.

Ubuntu Mytek Brooklyn DSD Setup Guide

Update August, 2018:

These instructions are still valid, but I've gotten direct DSD working as well.  Please visit my updated post here:

http://pqltd.blogspot.com/2018/08/mytek-brooklyn-and-manhattan-dsd-with.html

I'm happy to report that the Mytek Brooklyn DAC is not only great sounding but it's a real treat to set up with Ubuntu. Why would you care if you are an OSX or Windows person? Because it's really inexpensive to set up a music server with Ubuntu.  From under $100 Raspberri Pi 3 to my own mini box which was under $500. In all cases, an Ubuntu server competes very well with $2,000 or more dedicated streamers. 

One major feature that is missing from a lot of "audiophile" solutions is Android app support.  This is something we will overcome quite easily.

Also, since this is really a driverless installation, these instructions will probably work for any other modern USB 2.0 DAC that claims to be "driverless" for OSX and Linux. The catch will be in setting the right device (SL_SOUNDCARD, below) but that part is easy.

There is a known limitation. Currently Squeezelite only supports DoP, which is horribly bandwidth inefficient. DoP is DSD over PCM. It is not the same thing as direct or native DSD play.

It's not my code, but I'm working on it.

Introducton to the Logitech Media Server

There are a lot of Lunix music players out there, I'm going to go with the easiest one that gives me bit-perfect playback: Logitech Media Server 7.9 and Squeezelite.  You need at least 7.9 to get DSD player support as it's a fairly recent feature. I think it was introduced in 7.8, but I never tried it.

Originally developed for the highly regarded streamers created by Squeezebox and bought by Logitech, LMS has pretty much transcended Logitech AND Squeezebox and now has a robust and active Open Source development community. It's easy to use, convenient and free, and has Android and iPhone apps available. On Android I use Squeezer but on iPhone you can try the original controller app or SlimLibrary.

The one thing that is missing from LMS is a music management feature. It doesn't tag, organize, or do much else but play music. On the other hand, it does this VERY well and without complications. For this you'll have to find another application of which there are legion.

Also, LMS has fantastic and simple Internet radio discovery and playback. It's a real marvel in this manner for which on Linux I have no better alternative, nor do I want one.

Preparation

First, you must have Xubuntu/Ubuntu 14 or greater set up. I am on 16.04 LTS, but 14.04 LTS was identical.  If you have another Linux distro, see the link to the details, below. I will say in the past I used to run Fedora and LMS was one of the main reasons I quit it, so if that's your distro and this is difficult, you have my sympathies. May the gods have mercy on your heathen soul.

Decide where your music and playlists are going to be. I prefer to put public media in /opt, so:

cd /opt
sudo mkdir Music Playlist log
sudo chown bob:bob *

Replace "bob" with your user account name.  Notice we are creating a log directory as well.  This is where any messages from the player (squeezelite) will be put.

Server Installation

First, install LMS from the public repo.  There are only nightly builds but seem pretty stable. You'll find detailed instructions here.

I used this script:

url="http://www.mysqueezebox.com/update/?version=7.9.0&revision=1&geturl=1&os=deb"
latest_lms=$(wget -q -O - "$url")
mkdir -p ~/sources
cd ~/sources
wget $latest_lms
lms_deb=${latest_lms##*/}
sudo dpkg -i $lms_deb

Mine is just a little different because I don't like cluttering up the root directory. This will look for the latest build, download to ~/source and install it.

After it's installed LMS should be running and available on port 9000. So open up firefox and click here:

http://localhost:9000

Follow instructions on the screen.  You'll need to setup your directories. Click on the lower right of the page, "Settings." and setup your music and playlist folders. The only tricky part left in LMS now is to enable the DSD player.  Click on the "Plugins" tab and enable "DSDPlayer."

Very cool part of this is that DSD files will be automatically transcoded if your player doesn't support DSD directly. I have a Squeezebox Touch that does not, so that gets transcoded to 88.2kHz PCM automatically for me.

Make sure your music and playlist folders are universally readable.

Player Installation

OK, that's it for the server.  Now we need to get your PC to talk to the DAC. This is even easier since the Ubuntu distribution includes squeezelite. Let's install it:

sudo apt-get install squeezelite

I've tried the distro version with Ubuntu 14 and 16, and both work great with the Mytek Brooklyn. DSD and HiRes PCM played without a hitch on both so don't feel put out if you don't have the latest version. It's not worth the trouble to get the nightly or the source and build.

Make sure your DAC is now plugged in and turned on.

You'll find the configuration file for Squeezelite in /etc/default/squeezelite.

Here is how I configured it:

# The name for the squeezelite player:

SL_NAME="Brooklyn_DAC"

# ALSA output device:
SL_SOUNDCARD=""hw:CARD=DAC,DEV=0"

# Squeezebox server (Logitech Media Server):
# Uncomment the next line if you want to point squeezelite at the IP address of
# your squeezebox server. This is usually unnecessary as the server is
# automatically discovered.
#SB_SERVER_IP="192.168.x.y"

# Additional options to pass to squeezelite:
SB_EXTRA_ARGS="-D -f /opt/log/squeezelite.log -d all=info"

# As above, but without the log file

# SB_EXTRA_ARGS="-D"

The key part is the SL_SOUNDCARD setting. If you aren't using the Brooklyn, I'm not sure this will be right.  I found it by using "squeezelite -l" and looking for the entry that said :

USB Audio - Direct hardware device without any conversions

There were other entries that worked, but I found they would convert whatever I played to 48/32, no matter the original.

Another important switch is -D which enables playing of DSD without transcoding to PCM.  Without this switch your DAC will always get PCM.

You'll see two versions of SB_EXTRA_ARGS. The first adds a log file, but really once you've got it all working it's not very useful. Once you are happy it's working comment it, and uncomment the last line, then restart.

One last command and you should be running:

sudo service squeezelite restart

Assuming all went well, you should see "Brooklyn_DAC" (set by SL_NAME, above) appear as an available player in LMS on the top right.  Select it and play some music! If you have music but it's not showing up you need to scan your library. Click on the bottom right "Settings" and in the "Basic" tab check the media folder is correct, then click "Rescan" and your files should get imported quickly.

If you need to free up your DAC for movies, or games use the player OFF switch in the LMS screen on the top right.

Now of course it is time to download some good sounding DSD music, for which I highly recommend Blue Coast Records.


Happy Listening!

Meridian Quality Authenticated - It's Snake Oil, until it's not

MQA 

Update: After a lot of reading, MQA is NOT lossless. What a shame.
A full treatment is available from this post at Benchmark Media

In fact, it's such a good and thorough post that I'm going to delete the rest of my original posting and let it speak for itself. 

I will also add that you can listen to MQA files and compare them to the originals via the 2L test site here. 

After listening for a while with a Mytek Brooklyn DAC I can honestly say I hear no difference between an MQA file which claims 384k to the "merely" PCM 96k/24 bit files. 

Wyred4Sound Remedy - First Listening Impressions

I have hooked up the Remedy between my Logitech Squeezebox and my Audio Research DAC 8.  The only true jitter measurements I could find were for it's slightly newer cousin, the DSPre.  It seems that the DSPre is wildly sensitive to jitter, at least the measurements are, so I'm just going to assume the DAC 8 is the same or worse.

Most of my listening these days is to Internet radio, including Toronto Jazz FM 91, which in addition to having great programming also streams at 24/96.   The other station was KDFC 90.1, Bay Area Classical.

In the middle of this my power regulator has started to hum, so I can't do the remedy justice until I move it to a quieter location.

Still, here's what I think so far.

The remedy works much more noticeably with low-resolution stations, but it IS better. The sense of space inside the sound stage and the treble decay. Sometimes it feels worse though. During mass string crescendos the sound gets too complicated.  With Jazz 91 the improvements seem much less pronounced, but still there.

However this is really hard to gauge with radio.  I'll give this a better listen soon, when my biggest noise sources have been fixed.


Update August 28, 2016
So I had thought my impressions might greatly change, but they did not. I think this is a good tool but only for more source sensitive DAC's like I had (just sold it), the ARC DAC 8. That DAC played brilliantly when driven by an Ayre CD player, but when I brought it home it was much fussier.

With high resolution music (96k and above) I could honestly not tell if it was working or not, which I guess is a good thing. So, overall I would recommend this to clean up the sound of a mid-Fi CD player or Internet radio or inexpensive streamer like my Squeezebox Touch. 

I have however switched over to a Mytek Brooklyn DAC which sounds as good as the Wyred4Sound + ARC DAC 8 without the remedy in place, regardless of how it was driven.

I'm now driving the Brooklyn with a 2TB Linux streamer I built myself ($650) and it's very happy to play PCM, MQA and DSD from it.

Digital Audio - Upsampling and Oversampling Explained

Many types of digital sources, accessories and Digital to Analog Converters (DACs) provide some sort of sample data magic called oversampling or upsampling.  Put simply it means you end up with more digital data than you started with.

There are some benefits, but none of these methods truly gets you closer to the original music. They are all just ways of trying to make the experience more pleasant. Think of it as looking out your window with a screen. You may take a picture and find that you can see the screen itself in the image, or you can do some editing with Gimp or PhotoShop and remove it. The new image can't possibly contain more true to life data than you started with, but the picture should be much more pleasant to look at.

Many audiophiles have been led to believe that this kind of digital math can do things like you might see on the TV shows CSI or NCIS. Somehow four pixels on a grainy satellite image can be processed over and over again until the criminal's face is clearly visible. It's just not true.

Looking at it another way, the frequency response of up and oversampling does not change. A 44.1 kHz file is not going to have 30kHz created after 4x upsampling.  The frequency range and content density is unchanged. What may happen is that digital filtering becomes smoother and easier on the ears, or that jitter is improved somewhat by the use of higher data rates.

Differences Explained

Let's take original data.  Since digital music is always integer, I'll imagine two consecutive samples with convenient values of 24 and 28. Now lets see what happens at 4x up or oversampling. If the original data was 44kHz/16 bits the DAC will now see a sample rate of 176.4 kHz but the bit depth may or may not. So, just to be thorough, here is our original data:

• 24
• 28

at a sample rate of 44.1 kHz these two samples represent:

• 2 samples / 44,100  = 45 microseconds of music. 

 Remember that we are adding samples in between the time slots, so we don't want to stretch out our time, that would result in pitch changes. Instead we increase the rate (samples / second) at which we feed the DAC, keeping the pitch constant.
So, instead of 2 samples, we have 8, but with a new sample rate.  Lets redo the math:

• 8 samples / 176.4kHz = 45 microseconds of music.

Thhat's great, because if that didn't work the sound would be 4 times slower. :)

Oversampling

This is the oldest trick in the book. Almost immediately after CD players became commercially available oversampling became a buzz-word. I am no longer sure, but this may have only worked with so-called Delta-Sigma or 1-bit DAC's.

It's so simple you don't think it should work. Take a sample, and repeat it several times. It's that simple. It does not attempt to provide any more data but may shift some noise far above the Nyquist frequency.  No math is involved, just counting.  With 4x oversampling the DAC our orignal two samples become:

• 24
• 24
• 24
• 24
• 28
• 28
• 28
• 28

It's weird it helps, but it does. In fact, with oversampling, only 1 sample really matters at a time.

Upsampling

Bit Perfection

One of the objections to upsampling, is that the signal is no longer bit-perfect.  The DAC no longer gets the original facts, but the original facts, plus a lot more. That "lots more" is pure mathematical conjecture. However, there are some real benefits to be had.

Things get even more muddled when upsampling is used for ASRC, Asynchronous Sample Rate Conversion, but also more beneficial, as it's one of the best ways to reduce jitter.  More on that in a future post.

Technically and mathematically more challenging, there are two general approaches. To take the best advantage of this it's better if the bit depth increases beyond the original. So if the original was 16 bits, 24 or 32 bits will provide better resolution.  However remember that this doesn't really make it more true to life. It just makes some things easier to do and helps us keep more of our results. There are some VERY nice 32 bit DAC chips out there though, so taking full advantage of them may also get us much closer to true 24 bit resolution. That's a topic for someone else.

Linear Interpolation

Imagine two points on a chart. Draw a straight line between them. That's simple interpolation. It's no more complicated than simple algebra. Calculate the rise, divide it by the number of intervening samples, and add that much for each "new" sample. For linear interpolation, the sample rate converter needs to know two samples at a time in order to figure out the rate at which the intermediate samples should change.


Again, consider our original two samples, 24 and 28. The rate of change is 4/sample.  4/4 = 1. Now the DAC gets:

• 24 +1 =
• 25 +1 =
• 26 +1 =
• 27 +1 =
• 28

We'll just assume there's no bit-depth changes, or that in this case no extra resolution was required. Of course, I chose 24 and 28 to make the math here easy.

Spline

A much more advanced way to create more samples is by using what are called splines. Remember the "French Curve" tools you may have used in drawing school?

Technically you only need 2 samples for a spline, but the result is the same as linear interpolation, so we'll ignore that case. With spline math we take a number of samples, usually under 20,  to draw a much softer curve. Wadia was the first company I know of who introduced this concept. In this case it really helps to have more bits, as the extra bits help with more fine grained results. As you might imagine, the math and CPU power required is greatest for this example.

If this was floating point math our working data set would be:

• (nine samples before)
• 24.000
• 25.185
• 26.355
• 27.888
• 28.000
• (nine  samples after)

Remember that what's really going on is that the algorithm is taking more samples into account than our original two in order to fit the curve properly.  So why the third sample is 27.888 instead of 27.978 or 26.500 has to do with the nine samples in the original file before the first (24) and after the last (28) shown here. It is believed, without a lot of proof, that this method may provide the most natural resulting sound.

Are Splines Really Better?

Splines are very cool, but it may be argued, convincingly, that we are not doing much more than you could achieve with a capacitor and resistor with the proper time constants. In other words, it's a lot of math and hardware for what could be done with $2 or less in parts. The real potential benefit of this advanced though is in custom algorithms. You can be as creative as you want to in your algorithms.

What About Sound Quality? 

Personally I have come to believe that the analog output stages matter much more than interpolating algorithms and sample rates or bit depth but the devil is in the implementation details. As always, buy what you like, and what is most pleasant to your ears. Don't buy algorithms or chips. Buy results, and spend no money that isn't pleasing to you.