From the heart or the head

Most of us understand that we can connect to music from multiple angles: emotional, analytical, as a student, as a teacher, as a critic or just just casually. However we are connecting we can easily tell when we like something or we don’t and we don’t need any sort of training for that to occur.

Case in point one of my readers, Randy Coleman, sent me this note about his A/B testing of the NuWave DAC.

“As I switched back and forth between the CD inputs, my wife hollered at me from the other room, “Why is the music jumping?” I told her what I was doing and asked her to tell me whether A (DL III) or B (NuWave) sounded better. A self-described tone deaf, tin-eared music tolerator, she said B was dramatically better with every music sample. She used the terms “richer,” “more alive,” “more depth,” “less grainy,” and “sounds of individual instruments are clear.” Trust me, she’s never read a hi-fi magazine.”

No, she’s never read a hi fi magazine but she knows what she likes. Why? Because she wasn’t trying: her analytical mind was switched off and her emotional connection switched on.

Whenever I am put on the spot to evaluate something my immediate reaction is one of anxiety – I become guarded as it feels like I am on trial. It’s taken me years to learn how to relax under this stress and turn on the emotional “engine” in me.

As designers, however, we must be able to go in and out of the two states in order to relate what we get emotionally connected with to its root cause.

Tomorrow, learning to design by listening.

Paul Mcowan – PS Audio, Intl.

Trying too hard

When we first started PS Audio in the mid 1970′s Stan and I were hell bent for leather on getting as close as we could to our reference preamplifier’s sound, the Audio Research SP3A4 which is still a great sounding, musically satisfying design even to this day. We wanted to produce an affordable version of this fine preamplifier, one people like us could afford as the SP3 was, in those days, a lot of money.

As we got closer to the musicality of the AR, with our solid state design, I was still struggling to hear this “musicality”. Focusing as hard as I could on the quality of the cymbals, the pluck of the bass the tiny nuances of the voice I could certainly tell differences but I could not, for the life of me, tell which was more musical. Musicality in a piece of stereo equipment was a concept that really eluded me – yet seemed so obvious to the group of Audiophiles we knew at the time.

It’s tough being the one guy in the crowd that knew more about how things worked but less about how well they worked than anyone else.

I was always a music fan and went to as many live concerts as I could but honestly, those concerts never sounded even remotely the same as even the best stereo systems I had ever heard. Yet this live music seemed to be the standard by which everyone judged the performance quality of their stereos. It was baffling.

The big breakthrough for me came from my partner in PS, Stan Warren. Stan said something like “you’re simply trying too hard. You’re focusing on the gnats and gagging on the whales. Does one sound closer to actual musicians playing in the room and the other more like a hi fi system trying to duplicate the same?”

Surely it couldn’t be that simple. What Stan was telling me was something I have never forgotten to this day – to stop focusing on the little specific bits in a performance and step back to take in the whole experience – despite the fact the musicians never really sound like they are in the room (in the literal sense of the words) it is all relative.

It may seem obvious to many of you what I have just written and I am sure my initial ignorance applies only to me and a few like me. I get very literal at times – a tendency I see a lot in our engineers and programmers. The nerd in me was just struggling to look so closely at the details in order to find the whole.

We’ll look deeper tomorrow.

Paul McGowan – PS Audio, Intl.

Court and Spark

Back in the mid 1970′s when Stan and I were first starting PS Audio we both had a lot to learn: Stan electronics, me listening. Stan was the Audiophile while I was the nerd who knew how to design stuff and we both helped each other learn what was missing in our quest to build a high end audio company that could bring affordable equipment to Audiophiles.

At the time we had but one product: the original phono preamplifier. Our goal was to get as close as we could to the Audio Research SP3 preamplifier’s phono stage and that, I assure you, was a tall order. The SP3 has always been one of the most musical and enjoyable sounding preamplifers I have ever listened to: even to this day. It has a wonderful, big, rich and euphonic sound to it – accuracy not being part of its vocabulary – but musicality is over the top. Our competing design was more accurate, had better bass response, better measured response, yet sounded less like music than did the AR.

At first I had to rely completely on Stan’s judgment of the differences and that was a frustrating experience for me. I could hear the differences between the two quite plainly, but to my ears the extended bass response, crisper highs and smoother sound of the PS product was better than the Audio Research which sounded rather bloated and tubby to me. Stan assured me that one sounded musical and the other did not. It was a mystery but I trusted Stan’s ears and they were confirmed by every other Audiophile in the room.

After every listening session we’d discuss what our next steps were to get closer to our reference. For me it was like shooting in the dark because while I could hear every change we made I couldn’t really tell if it was Audiophile better or just different.

One of the most popular albums of the time was Joni Mitchell’s Court and Spark and we listened to tracks like Help Me and Free man in Paris until I think I knew every note and word by heart. I focused intently on the bass, the highs, the depth of the image till I thought my brain would hurt.

As we got closer to the Audio Research reference sound, a process that consumed nearly a year, I started to understand what it was Stan was hearing and I was not.

Tomorrow, we’ll share the rest of the story together.

Paul McGowan – PS Audio, Intl.

The art of listening

One of the best listeners I know is my wife Terri. She can sit down in the listening room and hear a track she likes, listen to the same thing on another piece of equipment and instantly knows which she likes better – and the test can be repeated time and again with the same results. The difference in equipment can be supremely subtle yet there’s no fooling her.

I, on the other hand, always hear the differences as well but struggle with “which is better” and tend to over think the choices, justify the reasoning and so on. I am sure many of you can relate to this situation and are always grateful when a third party walks in and declares the winner so easily – even if in the end you wind up not agreeing.

I thought we’d spend some time together on this subject: the art of listening. It’s a complex subject and one I bet we can both learn from as we explore it together.

Tomorrow I’ll share some of my first listening challenges as a designer.

Paul McGowan – PS Audio, Intl.

PS and DSD

Over the last few days we’ve been discussing the two main technologies in digital audio today: PCM and DSD. In my mind there’s no doubt that DSD is superior to PCM – if for no other reason than the fact it is simply closer to analog than anything I have ever heard. Properly implemented you’re not even aware of its presence and that’s how any piece of equipment or format in a high end audio chain should be: not there.

We’ve also learned that nearly all modern 24 bit and higher DACS are essentially PCM to DSD decoders and the final outputs of modern DACS is already DSD in nature.

So what’s the future of DACS in the PS Audio ecosystem? DSD, of course. We’re right in the middle of a long range, long term project to fundamentally change the nature of the DAC itself from that of a compromised PCM/DSD decoder to a purpose built DSD architecture that also accepts PCM without compromise. It means getting away from off-the-shelf DACS and chips and heading down a brave new frontier essentially alone – but that’s ok, we kind of thrive on that.

The fruits of these labors won’t be enjoyed for quite some time but in the scheme of things it won’t be that long. Perhaps sometime in 2014 we may see a new breed of product from some of the really great minds we have working on this now.

I am delighted to be along for the ride.

Paul McGowan – PS Audio, Intl.

So how come?

As we end our series on PCM and DSD I have been fielding a lot of questions concerning our products and the lack of a DSD input. A few of you have also asked why we don’t have an SACD player and so on.

Let me share with you a few of the issues going on and shed some light onto this subject. The first thing we should do is make a distinction between SACD and DSD. While they use the same format, SACD is a proprietary optical disc of Sony (not a format) and all SACD releases from Sony are copy protected as well as heavily guarded against getting a digital output of the original data under any circumstance.

Sony was a pioneer in the recording of and remastering of original high resolution audio material from artists. First introduced in 1999, which coincidentally was around the same time as Napster was giving the music industry fits and Steve Jobs was negotiating deals for legal download of music, Sony had to convince wary artists that they could offer 100% protection for the music. Remember, the music industry was upset about CD copying and MP3 sharing and along comes Sony asking for the right to distribute openly what was essentially the master tapes! CD copying is bad enough, in the eyes of the music industry, but to give any kind of access to the far superior master tapes was just nuts. It was a big challenge and frankly I tip my hat to the army of lawyers that pulled it off.

Each copy of a SACD has a unique ID and it has proven nearly impossible for anyone to crack the encryption scheme used to prevent copying. Legally you cannot make the raw unencrypted content of a SACD available in digitally. You may have seen SACD and Blu Ray players that have HDMI digital outputs that can go into a receiver or DAC, but I can assure you that the decoding of the encrypted data stream on the HDMI cable happens only inside the chip itself in the DAC. What’s in the HDMI data is encrypted and of no use to a DSD capable DAC.

Products like the Oppo output an unencrypted DSD stream but it is by no means the raw DSD stream – instead it is a truncated and reduced version that is not even close to the original. Many SACD players also output a digital audio stream that many believe is the raw data, but this is actually PCM from a second layer on the SACD that Sony included so if you bought the disc but didn’t have a SACD player you could still play the disc in your CD player.

Sony simply cannot ever allow the raw DSD data to be available because if it did then one could easily copy that data and that would violate all the agreements they have in place with the artists. You really cannot blame Sony for this, I am sure any lawyer would have a field day if they did.

With respect to a DSD capable DAC from PS Audio, let’s look at that tomorrow and then we’ll move on.

Paul McGowan – PS Audio, Intl.

Recording it all

There are about as many opinions on the sound quality of DSD vs. PCM as there are people who have them but the general consensus seems to favor DSD. I know from personal experience with my friend and mentor Gus Skinas, who has personally mastered perhaps a third of all SACDs’, that properly implemented DSD is hands down more analog like than PCM.

Part of the problem in this ongoing saga of standard CD/PCM vs. SACD/DSD technology is that there are very few studios that record directly onto DSD – most going PCM and then later mastering from PCM to DSD should the need arise. I think most of us would feel comfortable with the statement that there’s perhaps no advantage going from PCM to DSD and, in fact, most likely a loss.

The Sonoma DSD system as well as the Korg DSD recorder are probably the two main players in encoding DSD but, again, use of these two systems for original recordings is quite limited in the big world of recording studios. Great advocates of DSD recording, like Cookie Marenco of Blue Coast records, deserve a lot of kudos for furthering the art.

Another of the hopeful signs in favor of DSD’s superior sound quality to PCM is the conversion of original analog master tapes to DSD on the Sonoma system. Gus Skinas is one of the experts in this field and I can tell you from personal experience listening to a DSD master of an analog tape is nothing short of breathtaking.

Despite the fact that Sony has chosen to abandon it’s SACD format interest in DSD and SACD’s continues to gain strength and steam and we feel that’s a good thing.

Paul McGowan – PS Audio, Intl

Errors upon errors

How many errors must a DAC make to get it right? Well, in a classic PCM DAC, the answer is none. 16 bit DACS of long ago relied on a perfect conversion process without any errors to get it right. In fact, the whole idea of PCM audio is extreme accuracy the first time around without ever looking back. It is a very rigid system (although certainly not perfect).

DSD/SACD audio, is exactly the opposite. The process used in a 1-bit system, known as Delta Sigma (or Sigma Delta), relies on a very clever error feedback setup. This works by “testing the waters” one bit at a time and then comparing the results with the original signal to see if it got better or worse.

Yeah, I know, it sounds quite squirrely to just poke around until its right, but that is a very simple explanation of what the encoder does to generate the original DSD signal which, by the way, is not exactly what happens when we convert fixed PCM to DSD.

Let’s follow the path our signal takes on the way to becoming encoded.

Our analog signal comes in as a varying up and down voltage as we know from reading past posts in this series. There’s a very simple device called a comparator at the beginning. This device simply compares one voltage with another and tells you if one is higher or lower than the other: the comparator has two inputs. The first input is the music and the second input is feedback signal from the output of the DSD encoder.

If the output of the DSD encoder is low, relative to the input, then the encoder puts out a 1. If it looks again and it is still low, it puts out another 1. If it is too high, it puts out a 0. It does this over and over 64 times faster than a standard CD sample. So again, it just compares the input with the “output” of the encoder and gives out a 1 or a 0 to try and keep everything the same between the musical signal and the output signal.

In this way, it keeps correcting the digital signal until it matches as perfectly as the system allows to the input and it does this in a continuous stream of data without code and without any fixed steps of voltage as PCM does.

The one issue you have is a lot of noise in this process and to eliminate that a process known as noise shaping is employed – the math of which is far beyond anything I can understand.

Paul McGowan – PS Audio, Intl.

Timing is everything

Yesterday we discovered what 1-bit DSD looks like and how it’s fundamentally different from that of the standard CD format, PCM. I am going to spend a few posts helping folks understand a little better detail about DSD but today I wanted to make sure we all understand the importance of timing.

Timing is everything in digital audio. Regardless of the format, PCM or DSD, the accuracy of either the rate of samples we take or the accuracy of the 1-bit data coming in a stream on DSD plays a critical role in how the audio will sound when we play it back in our systems.

Timing issues are commonly known as jitter – a term I thought must have something to do with drinking too much coffee in the morning – but turns out to be simply differences in timing of the stream.

If you’ve been following along in our little series it should start to be clear how important jitter can be to the eventual output music. Consider that in a PCM system if the timing of the samples, which is supposed to happen 44 thousand times a second starts changing in the encoding process or the playback process. The system only works well if the timing is correct because, at the end of the day in both DSD and PCM they are timing based systems. Change the timing and the results will always be something unexpected.

Tomorrow we’ll start on the crazy process in 1-bit decoders of trying and failing to get it right.

Paul McGowan – PS Audio, Intl.

How dense are you?

Yesterday I surprised a few of you when we learned that the output of just about every modern DAC with 24 bit resolution is actually DSD, or Pulse Density Modulation, also commonly known as 1-bit. We understand that classic ladder type DACS reach a brick wall at about 16 bits because the accuracy of parts needed to extended beyond that range simply do not exist.

I would ask if any of you remember back to the beginning of this series when I told you that PCM (which is what CD’s work with) is a code decipherable only to a machine that understands that code – and DSD is not a code at all and your analog stereo system can play it back directly. That’s a big difference.

We learned that PCM measures the musical signal with a series of snapshots called samples and memorializes each sample as a multi-bit word. We reconstruct the samples by reversing the process and creating a stepped output which is identical to the input once we smooth out the little steps.

1-bit audio is very, very different. Today we’re going to simply give you an overview of how it works, which is a simple process, and then we’ll get into the nitty gritty of the process which can be a little disturbing to say the least. But let’s save the disturbing stuff for later once we’re all on the same page.

1-bit audio is simple to understand in concept. There are no samples, there are no words, there is no code. Instead there is a continuous streaming “train” of single identical bits that are either on or off. The more bits that are on, the higher the eventual output voltage becomes. The more bits that are off, the lower the eventual output voltage. We refer to this type of scheme as Pulse Density Modulation because when you have a greater number of on bits it appears as more densely populated. Here’s a picture that will help you visualize a 1-bit system.

Pulse density modulation 2 periods How dense are you?

Note the blue areas are on and the white areas are off. Also note the periodicity between single bits is identical. The red sine wave overlaid on this image shows the results of more bits or fewer bits. Where there are no on bits (all white) the sine wave is at its lowest point – lots of on bits and it’s at its highest point.

The speed of the bits is 64 times the sample rate of a CD and some DSD schemes run at 128 times faster than a CD.

Here’s the interesting part of this: if you take a DSD stream and run it through a simple analog lowpass filter to smooth out the on/off transitions, you get music! This is amazing considering that if you do the same with PCM you get only noise.

DSD is a lot closer to analog than PCM ever thought to be.

Paul McGowan – PS Audio, Intl.