Mojo Audio: Latest News

Music Streaming without a Rat’s Nest

Wed, 18 Mar 2026 15:24:47 +0000

Introduction:

Streaming music has become the go-to source for most music lovers. Millions of songs from every genre cross-referenced in a user-friendly database.

What’s not to like?

As with every aspect of high-end audio, music streaming has its challenges and compromises. Because of this every year more companies are offering “magic boxes” to improve your music streaming experience.

Ethernet filters, USB filters, Ethernet switches, optical switches, optical isolators, Ethernet reclockers, USB reclockers, and master clocks. Most of these require their own AC power source and power cable. And then you need to connect one box to another using an expensive digital signal cable.

The result is a rat’s nest of cables and components.

Can you get state-of-the-art music streaming without all those devices?

No.

But you can put all of them into one chassis, get higher performance, have more versatility, and save money.

This blog will show you how to build your own “One Box Wonder” that will rival the best-of-the-best networked streaming systems for a fraction of the price.

Why own a custom streamer vs an appliance?

If you want the simplest and most user friendly streamer, then you may want to opt for a streamer appliance.

The only drawback with an appliance is they are what they are: their hardware and software are not upgradable. And there is no guarantee that the manufacturer will repair or support it once it’s out of warranty.

Next thing you know you want to upgrade your system’s performance and you’re either having to buy a new streaming appliance or adding “magic boxes” to the input and output.

Now your simple inexpensive solution is no longer quite so simple or inexpensive.

On the other hand, with our "One Box Wonder" you can not only get higher performance for less money, you can also have a streamer that’s relatively obsolete proof.

Linux and Windows compatible.
Use any player software or digital signal processing software.
Integrated USB, Ethernet, TOSLINK, SFP optical, I2S, or S/PDIF cards.
Integrated switches, modems, reclockers, and master clocks.
Easily reconfigurable, upgradable, and repairable.

Versatile, inexpensive, and obsolete proof: could you ask for a better combination?

Why is a single chassis better?

When using multiple music streaming components, between each component there is a sending chip, a receiving chip, a cable, and connectors. Multiple power cables and AC distribution are required. And because you are using multiple power supplies potential ground loops can be generated.

All these things degrade musical performance.

When all these components are integrated into one chassis with one power supply you only need one AC receptacle and one AC power cable and you eliminate potential ground loops. Better still, all the components communicate through the main data buss on the computer’s motherboard instead of through a rat’s nest of expensive data cables.

Integrating everything into one chassis improves performance and minimizes size, cost, and complexity.

Why use the Streacom FC10 Alpha chassis?

The Streacom FC10 Alpha is a beautiful high-quality all-aluminum chassis. The FC10 is large enough to hold all the required components, it’s available in either black or silver, and it has two slots for PCI cards.

Our design uses the right heat sink for the CPU and the left heat sink for our integrated power supply.

Can you build any type of streamer inside of this chassis with our integrated power supply?

No.

With our integrated power supply you have only enough space for a mini-ITX motherboard.

And the total current our power supply can provide is 8A peak and 5A continuous. So if you need insane amounts of processing power you’ll need to add an external power supply for your CPU.

But our integrated power supply has enough current to power a motherboard, 35w CPU, SSD, Ethernet card, USB card, and master clock. And more than enough current to power the same drives, cards, and clocks, and a high efficiency motherboard with an embedded CPU.

Unless you are using it for a professional recording studio or intend to use real-time digital signal processing (DSP), real-time music format conversion, or insane amounts of upsampling, a high-efficiency motherboard with an embedded CPU has more than enough processing power.

NOTE: a high efficiency motherboard with an embedded CPU would barely make the chassis warm, whereas a 35w CPU would heat up the FC10 chassis quite a bit.

A 35w CPU may give you more software options but all those DSP processes will degrade performance. The less you mess with the digital signal the more subtlety and nuance you will retain in your music.

If you need extreme processing power another option would be to use our integrated power supply for your drives, cards, and clocks and add an external power supply for your CPU.

With this option you could have up to a 95w CPU and you would still have relatively low heat.

The integrated Illuminati FC10 power supply:

Our “One Box Wonder” has an integrated LC choke input power supply: the largest, heaviest, least efficient, and most expensive power supply typology you can use.

LC choke-input power supplies have the highest impedance to AC, lowest impedance to DC, 50% the crest factor and heat of a capacitive power supply and require a power transformer that is 50% larger.

No small difference.

This translates to a power supply that has the lowest noise, the most effortless dynamic performance, the most isolation from AC noise, and the most durability of any power supply typology.

Our FC10 power supply uses the best-of-the-best component parts including:

Lundahl C-core balanced chokes.
Silicon Carbide zero-recovery Schottky diodes.
5X Belleson ultralow-noise ultrahigh-dynamic regulators.
Low ESR long-life organic polymer capacitors.
Furutech IEC AC input connector.

In addition, the four accessory power supply outputs use a standard Molex 4-pin PATA connector allowing for ease of connectivity to a variety of cards, drives, and clocks.

The 200VA toroidal power transformer is fully shielded within a steel cover. And the Lundahl choke is shielded behind a ¼” thick aluminum wall laminated with copper foil and advanced ERS paper shielding.

There are few if any power supplies that can provide this clean, this effortless, or this isolated power.

How is the “One Box Wonder” available?

We offer our “One Box Wonder” with four DIY options and several turn-key options to meet the needs of music lovers with different systems, different budgets, and different degrees of technical expertise:

Illuminati FC10 power supply kit ala carte ($1,499).
FC10 chassis with the power supply kit installed ($2,499).
Bare bones streamer with embedded CPU mobo ($2,999).
Bare bones streamer with 35w i7 CPU and mobo ($3,499).
Turn-key custom music streamer ($6,999 and up).

NOTE: with the bare bones options you get a fully tested chassis, power supply, and motherboard, but there are no drives, cards, or clocks. DIY options are available in a variety of voltages to meet your specifications.

We also offer remanufacturing services that allow you to send us your boards, drives, cards, and clocks, for us to install into the FC10 chassis.

What other components can be used?

Any mini-ITX motherboard from companies like Mitac, ASRock, Asus, and Supermicro can be used. But since the integrated power supply can only supply 5A of continuous current, the current requirements of the motherboard, CPU, and other components, must be considered.

We recommend no more than a 35W CPU. You could certainly use up to a 95W CPU, but that would mean you would need to use an external power supply for your CPU in addition to our integrated power supply.

One of our favorite high efficiency motherboards with embedded CPU is the ASRock IMB-154. The embedded Intel Quad-Core Celeron N3160 Braswell processor has more than enough power to stream music from the internet, play music files from your SSD, and do modest upsampling. And the incredibly efficient 7w total power consumption minimizes heat and noise.

The IMB-154 has two Ethernet ports, four USB ports, HDMI port, PCIe slot, and can integrate mini SATA and mini PCIe cards. Highly recommended.

If you require more processing power to run DSP software, one of our favorite combinations is the Mitac PH12SI-D motherboard combined with the Intel i7-7700T, 4-core, 3.8Ghz CPU. This high-efficiency single-voltage industrial motherboard and CPU combination has the processing power you’ll need to perform the most intensive DSP applications.

The Mitac PH12SI-D has two Ethernet ports, four USB ports, HDMI port, PCIe slot, and can use mini PCIe and half-size mini PCIe cards.

With any motherboard you can integrate high-performance USB, Ethernet, S/PDIF, I2S, or SFP optical PCIe cards from companies like JCAT, SOtM, Sonare, M2 Tech, Pink Faun, After Dark, Matrix, RME, and Lynx.

For USB cards, Ethernet cards, and master clocks, we recommend JCAT products both for their exceptional performance and because they run on the 5V power supplied by a standard 4-pin PATA connector.

A unique feature of the JCAT EVO cards is that by simply moving a jumper you can go from their integrated high-performance OCXO clock to a higher performance OCXO master clock. And one JCAT master clock module can simultaneously feed two of their USB or Ethernet cards. Quite a nice upgrade.

If you want to use components other than those we suggest you can order the FC10 power supply with the motherboard configured as anything from 12V-19V and with each of the four accessory power supplies configured as anything from 5V-12V. Contact us with your specifications.

Getting started:

For most people we recommend purchasing our power supply kit already installed in an FC10 chassis. This would allow any geek or tech to build their own “One Box Wonder” without any drilling or soldering.

For those of you with a bit more DIY skills, the following describes how to install our Illuminati FC10 power supply kit into your Streacom FC10 Alpha chassis.

Read through the Streacom FC10 User Guide. Unpack your Streacom FC10 Alpha chassis and confirm that you have all the parts. Parts are clearly outlined in the user guide.

Confirm your Illuminati FC10 Power Supply Kit contains the following:

Power supply module on L bracket.
Toroidal power transformer.
Transformer bottom pad, top pad, and clamp.
Transformer shielding cover.
Lundahl C-Core choke.
Choke shield/SSD drive mounting plate.
Adhesive backed copper foil and ERS paper.
Furutech IEC AC power inlet.
Motherboard DC power cable (not pictured).
Three syringes of thermal paste.
Bag of assorted hardware.

We offer an optional high-performance custom DC power cable kit that contains all the cables you will need to power two PCIe cards, one clock, and one SSD. Our high-performance PATA/SATA cables are made from cryogenically treated 19AWG Kimber VariStrand wire. Our custom cables are the exact lengths required as well as a nice performance upgrade over standard computer PATA/SATA cables.

Testing the Power Supply:

It’s always a good idea to test the power supply before assembly.

Plug everything together:

Clamp wires from power supply module into AC inlet.
Plug in power transformer’s in and out wire harnesses.
Plug in the choke.

NOTE: carefully follow the photo below and put the correct wires and the correct color wire harness into the correct connector on the power supply module. Doing this incorrectly will burn out the power supply.

To check the voltages on the power supply module you will need a volt-ohm meter. The photo below shows the voltages for our power supply module configured for a 12V motherboard and four 12V-0V-0V-5V Molex 4-pin PATA accessory ports.

Carefully plug a power cable into the AC inlet and check the DC output voltages.

CAUTION: shorting out the "G" and "5V" or "12V" pins will blow out the power supply module. When you put the black lead from your volt-ohm meter on the "G" pin and the red lead on the "5V" or "12V" pin be careful to not let the probes touch each other.

NOTE: for SSDs and most audiophile PCIe cards only the 5V on the 4-pin PATA connector is required. Our standard configurations does not have the 12V pin of the 4-pin connectors active. For custom configurations there are jumpers on the 12V pins that would allow them to be activated.

Once you’ve confirmed that you have all the chassis and power supply components and confirmed the power supply is working properly you are ready to begin the installation process.

Strip down the FC10 chassis:

Before beginning the installation, you’ll need to remove the following parts from your Streacom FC10 chassis:

Drive tray.
DC power jack AC inlet cover plate.
PCI expansion slot cover plates.
Auxiliary USB cables and connectors (optional but recommended).

NOTE: standoffs for mini-ITX motherboards are preinstalled in the FC10 chassis.

Mounting Holes:

The FC10 power supply kit uses several of the existing holes and some of the hardware that comes with the FC10 chassis. Some of the existing holes need to be enlarged and some additional holes need to be drilled.

To drill and enlarge the mounting holes you will need the following:

Electric hand drill.
Center Punch.
1/8” metal drill bit.
3/16” metal drill bit.
7/16” metal drill bit.
12” ruler.
Pencil.
Small piece of ¾” thick wood.

We engineered this kit with plenty of tolerance so that the average person with hand tools can install it. The bolt heads and washers will cover less than perfect drilling. Additional drill bits may be required to enlarge holes if your drilling is not done accurately.

Identifying and Marking the Holes:

To avoid scratching your chassis we recommend putting the chassis on a clean sheet of thin packing foam or a cotton towel while you are working on it.

Existing holes to be used are circled in red:

To mount the power transformer, you need to mark and drill additional holes:

Put the power supply module inside of the chassis with the L bracket touching the left heat sink.
Put one mounting bolt through the chassis to prevent the power supply module from shifting.
Put transformer cover in-between the base of the L bracket and the motherboard standoff.
Align the front left hole of the transformer cover to the existing chassis hole.
Put one bolt through existing hole to assure cover cannot shift while marking new holes.
Make sure cover aligns squarely to the front of the chassis and the L bracket.
Use your pencil to mark the other three corner holes for the transformer cover.
Be careful to cleanly mark the three additional corner holes without shifting the cover.
Remove transformer cover and power supply module from the chassis before drilling.

Now you are ready to mark the center hole to mount the power transformer. You are going to create an X at the center between the four corner holes:

Place your ruler diagonally from one corner hole to the next.
Align the ruler to the center of both circular holes before marking.
Mark a small line across the center between the two corner holes.
Rotate ruler diagonally and align it with the other two corner holes.
Mark a small line across the center between the two corner holes.

You’ll need to put a center punch at the center of each drill hole marking to hold the tip of your drill bit in proper alignment during drilling. This is one of the most critical operations. If your center punch is off your drill holes will not align properly. Not doing the center punching accurately is the #1 reason for needing to enlarge the holes in order to make parts properly align.

Drilling Additional Holes:

Put the piece of ¾” thick wood under the chassis below whatever hole you are drilling. This will prevent the drill bit from distorting the bottom of the chassis as it passes through.

IMPORTANT: before going from one drilling operation to the next brush or vacuum the surface clean of any drill swarf that could scratch your chassis.

Use the 3/16” drill bit to enlarge the choke mounting holes.
Use the 3/16” drill bit for the transformer cover’s corner holes.
Use the 7/16” drill bit for the transformer center hole.
Drill a 1/8” pilot hole before using the 7/16” drill bit.

After drilling the holes align each part to their respective holes to assure the parts will mount correctly. If the holes seem to visually align properly, loosely install the mounting hardware to confirm that there is enough tolerance for the parts to mount flush with the base of the chassis.

If the holes do not align correctly you will need to enlarge one or more of the holes with a larger drill bit. Enlarge one hole at a time and check to see if the part aligns. Start with a 1/64” larger drill bit and go up in 1/64” increments.

Mounting parts in the chassis:

When mounting parts with multiple mounting holes it is advisable to put in all bolts loosely first, and then once they are all equalized in spacing, carefully tighten each bolt one at a time.

Mount items into the chassis in the following order:

IEC AC inlet.
Choke.
Choke shield/SSD mount.
Power supply module.
Power transformer.
Power transformer cover.

Mounting the IEC AC inlet:

Slide the IEC AC inlet into the hole in the rear of the chassis. It should fit snugly. Use the two 4-40 flat head bolts and locking nuts provided in the power supply kit. Insert both bolts and hand tighten both nuts. Tighten the bolts using a Philips head screwdriver while holding the nuts in place using a ¼” open end wrench. Do not overtighten or you will crack the plastic.

Mounting the choke:

The choke is mounted using two M4 round head bolts provided in the power supply kit. Align the choke with the wire cable facing the front of the chassis. Insert the tips of both bolts and align the choke before threading them in completely and tightening them.

Mounting the choke shield/SSD mounting plate:

Before you can mount the choke shield/SSD mounting plate you need to mount your SSD.

And before you can mount your SSD you need to laminate the copper foil and ERS paper.

Copper foil laminates to the flat side of the L bracket.
Peel the backing off the copper foil.
Do not allow the copper foil to curl up and stick to itself.
Align the copper foil to the top edge and overlap the vertical edges.
Adhere and smooth the copper foil using a plastic spackling knife.
Remove the backing from the ERS Paper.
Center the ERS paper between the SSD mounting holes.
Apply the ERS paper on top of the copper foil.
Optional: ERS paper for on top of SSD - adhere over center.
Trim excess foil and ERS paper from around plate and SSD.
Poke copper foil out from over SSD mounting holes.

Now you are ready to mount your SSD to the plate using the round head M3 screws supplied in the power supply kit. Orient the SSD connector plugs towards the center of the chassis so that the SATA cable from the power supply module will reach it.

Now you're ready to mount the choke shield/SSD mounting plate.

Use three of the M3 x 8mm flat head bolts that come with the FC10 chassis to mount the shield/SSD plate. Align the holes in the shield/SSD plate with the holes in the base of chassis with the L facing the rear of the chassis. That would put the SSD on the same side as the motherboard and the opposite side as the choke. Partially insert all three bolts and align them before tightening.

Mounting the power supply module:

The power supply module mounts to the heat sink on the left side of the chassis using existing holes and four of the M3 x 8mm flathead bolts that come with the chassis.

Place the power supply module into the chassis and align it as shown. Put one of the mounting bolts into the power supply module to hold it in place. Use your pencil to trace the edges of the L bracket. These alignment marks will make it easier to align the power supply module when the L bracket’s upright is covered with thermal paste during final assembly.

Remove the power supply module from the chassis. Apply an even layer of thermal paste on the upright of the L bracket. Do not get thermal paste too close to edges and do not cover threaded holes or it will squish out when bolts are tightened. Smooth thermal paste with a stiff item, such as a small piece of cardboard or plastic spackling knife.

NOTE: thermal paste is quite sticky, not water soluble, and will stain anything it touches. We recommend wearing disposable gloves and using a disposable item like a small piece of cardboard to spread it. To remove excess we recommend using a paper towel soaked with alcohol.

Carefully place the power supply module near where it will be bolted into place. Align the edges of the L bracket to the pencil marks. Slide the L bracket toward the side of the chassis and stick it in place using the thermal paste. You may need to move the L bracket slightly forward or backward to get the first bolt to align. Align and insert the tips of all four bolts before tightening them.

Mounting the power transformer and cover:

The last parts to mount are the power transformer and transformer cover.

Initially only hand tighten the power transformer bolt so that it can be centered and rotated to optimally slack the in and out wire harnesses. Do not fully tighten the transformer bolt with a wrench until you are certain that it is centered and the in and out wire harnesses are evenly slacked.

Follow the sequence below to install the power transformer, pads, and clamp:

Put the chassis on a soft foam or cotton pad laying on it's face plate.
Slide the transformer mounting bolt up through the bottom of the chassis.
Put the transformer bottom pad over the transformer mounting bolt.
Slide the bottom pad down so that it is flush with the base of the chassis.
Put the power transformer over the bolt.
Put the top transformer pad over the bolt.
Put the transformer clamp over the bolt.
Slide the top pad and clamp flush with the transformer.
Put the transformer bolt's washer and lock-washer over the bolt.
Thread the nut over the tip of the bolt to hold everything in place.
Center the transformer, pads, and clamp before tightening the nut.
Hand tighten nut bringing the above "sandwich" of parts together.
Center all of the above over the transformer bolt hole before tightening.

Once the transformer is loosely mounted plug the in and out wire harnesses into the power supply module so that you can evenly slack both wire harnesses before installing the cover.

NOTE: there is a slot at the base of the transformer cover on one side. Orient this slot facing the rear of the chassis to allow the wire harnesses to fit through the slot.

Place the cover over the transformer and press it flush with the base of the chassis. Rotate and slide the cover so that all four corner holes align. Make certain that the in and out wire harnesses are evenly slacked. You may need to take the transformer cover off more than once to get the optimal rotation.

Once you are certain that all four corner holes on the transformer cover align perfectly and that the in and out wire harnesses are evenly slacked you can remove the transformer cover and tighten the center bolt of the power transformer using a ½” wrench.

IMPORTANT: make certain that the transformer’s wires are not being crushed or pinched by the cover. This can cause the transformer to short out, burn out, and generally make you have a bad day.

Installing the transformer cover:

Orient the slot in the base of the transformer cover towards the rear of the chassis.
Orient the wire harnesses to extend through the slot.
Put one of the washers over each bolt before inserting into the bottom of the chassis.
One washer goes on the bottom of the chassis and one on top of the cover.
One at a time Insert a bolt through the bottom of the chassis.
Each bolt goes through the base of the chassis and through one corner of the cover.
Put another washer over the tip of bolt on top of the cover.
Put a nut over the tip of each bolt and hand tighten to hold everything in place.
When all four bolts are in place with washers and nuts check wire harnesses.
Make certain no wires are being pinched by the cover.
Plug the in and out wire harnesses into the power supply module.
Assure wire harnesses are evenly slacked before tightening the corner bolts.
If wire harnesses aren't evenly slacked remove cover and rotate transformer.
Make certain none of the wires are being pinched before tightening.

Plug everything together:

IMPORTANT: make certain to plug the correct wires into the correct plugs or you will cause the power supply to short out, burn out, and generally make you have a bad day.

Make certain to plug the in and out power transformer wire harnesses into the correct plugs.
Before inserting wires into the AC inlet carefully twist strands and fold them onto themselves.
This creates a thicker bundle of wire which will make it easier to get a good tight clamp on the wires.
Make certain to insert the input wires from power supply module into correct holes in the AC inlet.
The purple ground wire coming from transformer goes into the center ground clamp of the AC inlet.
Make certain to plug the choke and motherboard DC power cables into the correct connectors.
Before plugging in the choke evenly twist the wires together.
Route twisted wires along front of chassis behind face plate and around transformer cover.
Doing any of the above incorrectly can cause the power supply to burn out as soon as you plug it in.

Testing the Power Supply (Again):

It’s always a good idea to test the power supply before installing the motherboard, cards, and clocks.

Make certain all the outputs are the correct voltage(s) before connecting any DC power cables.

Carefully plug a power cable into the AC inlet and check the DC output voltages.

Once you’ve confirmed that the power supply is working properly you are ready to install the motherboard, cards, and clocks.

Final Assembly:

Follow the manufacturer’s instructions for your motherboard and for the specific cards and clocks you are using. One of the best things about our Illuminati FC10 power supply is that you can use standard PATA Molex 4-pin computer power cables.

We recommend using PATA and PATA to SATA cables of the optimal lengths.

If you are using standard computer PATA and SATA power cables the optimal lengths are as follows:

PATA to PATA cables to cards and clock: 6” long.
PATA to SATA cables to cards and clock: 6” long.
PATA to SATA cable to SSD: 12” long.

The motherboard DC power cable can be ordered with a 4-pin ATX power plug that fits most single voltage motherboards like the ones we recommend.

If you are using a single-voltage to 24-pin ATX nano converter module you can also order the motherboard power cable with a 5.5mm x 2.5mm barrel plug.

Custom motherboard power cables are available by request. Price varies.

Before installing the motherboard’s DC power cable twist the wires evenly down the length. Plug the motherboard cable into the power supply and lay the twisted pair of wires along the front of the chassis behind the face plate and down the right side of the chassis.

NOTE: if you are using a CPU that is heat sunk to the right side of the chassis you may need to navigate the wires around the FC10’s heat pipes.

After plugging in all the power and data cables according to the manufacturer’s instructions you are now ready to test your streamer using a bootable flash drive. Only once you are certain the motherboard, CPU, RAM, SSD, cards, and clocks are working correctly should you install your operating system and player software onto the SSD.

For optimal performance and minimal problems we recommend the Euphony Stylus RAM-root Linux and music player software. Euphony Stylus can embed Roon, HQ Player, Tidal, Qobuz, YouTube, and other software. But for optimal sound quality we recommend using Euphony Stylus by itself.

NOTE: for optimal sound quality we recommend playing music from your SSD as opposed to streaming it from the internet. Most music streaming services will allow you to download your favorite music so that you can play them directly from your SSD.

IMPORTANT: Euphony Stylus has advanced license security. This means that once it is installed you cannot change any hardware without it locking up.

If you are using Euphony Stylus and you need to change any hardware in your streamer you need to contact customer support and have them unlock your license so that you can do a clean installation.

This is why thoroughly testing every component in your streamer while booted from a flash drive is so important prior to doing a permanent installation on your SSD.

Enjoy your “One Box Wonder” streamer!

10 Ways to Optimize Music Streaming

Mon, 11 Mar 2024 14:36:29 +0000

UPDATED: 4.1.26

Introduction:

Since 2015 over 12X the number of people have subscribed to paid streaming services. With over 400 million songs available online, the convenience of streaming has made it the #1 choice for most music lovers.

Over 57% of all revenues in the music industry come from paid music streaming services. In 2025 this exceeded $22 billion and it is expected to exceed $28 billion in 2026.

Nearly 33% of those revenues came from Spotify, nearly 13% from Apple, over 11% from Amazon, and almost 10% from YouTube. Interestingly enough, the revenues from audiophile services like Tidal and Qobuz don't even make up 2% of the total market.

With high-speed internet being the norm, if done correctly, the sound quality of streaming music can be almost indistinguishable from the same recordings played from a local library. Every Link in the Chain Matters. This blog will help you identify the weak links in your signal path so that you can get the best possible sound quality streaming your favorite music.

#1 Music Streaming Services:

Different streaming services have more of a selection of certain obscure genres of music than others. And for many of us the music selection is all that matters. For those of us who find most streaming services have most of the music we like to listen to, and who are trying to get the best sound quality, I recommend not falling for the trendy debates as to which streaming service sounds empirically better than another.

The sound quality of a specific streaming service in your home has more to do with just the quality of the recording and the specific hardware being used by that streaming service. Different services can have significantly different sound quality from one geographic location to another, from one internet service provider (ISP) to another, and even from one time of day to another.

I’ll get into that whole time of day thing when I discuss ISPs. For now let’s focus on streaming services.

Though many people find audiophile services like Tidal and Qobuz sound better than mass-market services like Spotify, Apple, or Amazon, that's not always the case. And though you may read definitively in this or that forum that one service sounds better than another, there are several factors that contributed to these often contradictory points of view.

I wouldn't recommend believing anyone who claims that one streaming service sounds empirically better than another. Rather I would suggest that you compare different streaming services in your home at different times of day and then draw your own conclusions.

As I explain some of the other factors that can contribute to the sound quality of streamed music you will begin to understand why sound quality is far from just a matter of choosing the right streaming service.

#2 Internet Service Providers:

Though higher download speeds may seem to be important when selecting an ISP, it is actually one of the least important factors. Modern ISPs have more than enough bandwidth to handle HD music streaming. Consider that all ISPs can handle HD movies which contain the significantly larger video portion along with multiple channels of HD audio. By comparison two channels of HD music use relatively little bandwidth.

Though things like the quality of an ISPs infrastructure and the consistency of their performance during different times of the day are important, what is significantly more important is the specific brand and type of hardware each ISP can use in your home. Some use better hardware than others. Some have the option to use various brands and types of aftermarket hardware. Some can be upgraded with a plug-and-play power supply upgrade. And others can do few or none of these things.

Let's start with what takes place outside of your home. The music travels quite a long distance between your streaming service and your home. Consider things that degrade performance such as how much their data lines are being subjected to EMI or RFI. And consider factors like corruption of the AC power used by their distribution nodes. In some neighborhoods DSL could sound better than optical. Or any ISP could sound better or worse during different times of the day.

The infrastructure between you and a streaming service and factors in your specific neighborhood may explain why you may prefer the sound of one ISP or one streaming service over another. Consider one streaming service might be west of your city that is all farmland and the other might be east of your city that is all industrial. It's not the streaming service but rather the route the music is traveling that cause the difference in performance. This is the reason why experienced audiophiles may have totally different opinions as to the sound quality of different ISPs and different streaming services.

Now let’s talk about things inside of your home. The performance of your modem, router, and their power supplies makes, a significant difference in sound quality. One ISP may have a significant advantage over another simply because there are more options to use higher quality modems, routers, and power supplies with their service. Whereas with some services you are locked into hardware that is powered with an internal switch-mode power supply (SMPS). Having the ability to upgrade the "wall wart" SMPS of your modem/router with a plug-and-play linear power supply (LPS) is one of the most significant things you can do to improve music streaming performance.

I’ll get into modem, router, and power supply upgrades later in this blog. Let’s get back to ISPs.

Another significant consideration and one of the more undefinable factors is the bandwidth in your area. Cable internet is a “party line” where you're sharing bandwidth with your neighbors. In some neighborhoods, such as older neighborhoods with old infrastructure or high-density apartment complexes or residential neighborhoods that share bandwidth with commercial areas, you may hear a notable difference in sound quality during different times of the day.

Read the fine print in the contract from your ISP: it clearly states “up to...” a certain download and upload speed and never guarantees those numbers.

I used to live in a place where low performance DSL sounded better than high-speed cable internet. In that neighborhood cable internet notably slowed down and sound quality suffered from 2pm when kids got out of school to about 10pm when most people went to sleep. And when I moved to a different neighborhood only a few miles away the same ISPs had totally different performance.

#3 Local Area Networks:

Let me start by mentioning the biggest of all music streaming faux pas: streaming through the Wi-Fi on your Local Area Network (LAN). Simply don’t do it.

I’ve run into quite a few audiophiles who had systems that cost more than their cars who didn’t realize how much sound quality they were losing because they were streaming through Wi-Fi. One of the biggest improvements they experienced was when they paid a modest fee to have their ISP relocate their modem to a room that would allow them to use an Ethernet connection to their streamer.

Now for hardwired LAN issues...

Though the download speed of your ISPs service and modem may be more than enough to handle your kids playing online video games and your wife watching Netflix while you’re streaming music, the act of your modem “juggling” data streams will degrade sound quality.

Even if you live alone you don’t want to download anything while you’re doing critical listening. And if you don’t live alone, you may want to consider a dedicated modem just for streaming music. And no matter how much isolation you may have you can always improve sound quality with better hardware.

#4 Modems, Routers, and Wi-Fi:

Most ISPs offer a single component that performs the functions of a modem, router, and Wi-Fi. None of these components are audiophile grade. Often the components your ISP provides can be upgraded with consumer or commercial quality components. You may hear how using commercial grade networking components improved a person's sound quality. Yes, commercial components will often sound better than the stock "all-in-one" modem/router/WiFi you got from your ISP. But in most cases they won't sound as good as even the cheapest consumer grade components that have been upgraded with an linear power supply (LPS).

Commercial networking components have higher bandwidth but that's not required to stream music. Aside from the stratospheric price another problem with commercial components is they have integrated SMPS. In order to upgrade the SMPS you'd need to perform surgery to remove and bypass it. And if you modify the power supply you'll void the warranty. I'm not saying this wouldn't be worth the time, trouble, and expense, just that as with most things in high-end audio, this would be another case of diminishing returns.

There are modest priced consumer quality components that are known for providing excellent sound quality. Since these components change from year to year, and since the compatibility of these components differ from ISP to ISP, this would be something you would need to research on audiophile forums.

But one thing that always holds true is to isolate your wired and your wireless components. Using a consumer grade modem or modem/router without Wi-Fi and plugging in a dedicated Wi-Fi component is always recommended. And always make sure to keep a bit of distance between your wireless and your wired components and your digital and analog components.

Here's a great bang-for-the-buck upgrade: even if you don't upgrade any components in your ISPs stock modem/router/WiFi but just plug the stock "wall wart" SMPS into an extension cord or power strip that is moved a distance from your analog components it will often improve sound quality.

There are also modest priced audiophile switches from companies like LHY. Some of these networking components not only improve the sound quality of Ethernet but also have SFP optical that would allow you to run optical cable in your home. Some are optically isolated or convert Ethernet to optical. A nice benefit of optical cable is that it is not susceptible to EMI and RFI noise. Ethernet cable can act as an antenna and pick up all sorts of EMI and RFI noise. Or the cables could pass by transformers or compressors that radiate noise. In some homes this makes a huge difference. In other homes the difference is not audible. Don't just assume that optical cable will sound better. It is quite location and infrastructure dependent.

Another nice option with these SFP optical devices would be to interface them with a high-performance network audio adapter like the Optical Rendu. No need for a dedicated optical interface or a server/streamer with an optical card.

Upgrading the "wall wart" or internal SMPS with an LPS will make the most notable improvement in sound quality. For most people I would recommend using modest priced consumer grade components upgraded with a plug-and-play LPS. This wouldn't require any technical expertise and wouldn't void the warranty.

#5 Power Supplies:

The “wall wart” and internal SMPS that come with most modems, routers, switches, NAS drives, reclockers, and mini computers, are the weakest links in your digital signal path. These SMPS create “hash noise” that get into the ground and radiant noise that pollute other components in your system. Even using the most basic LPS, like an SBooster, will significantly improve sound quality.

Of course simply isolating these SMPS from your analog components can make an audible improvement. Using different dedicated AC lines or different AC filters and keeping these SMPS or components that contain SMPS physically away from your analog components will minimize the effect of EMI, RFI, and hash noise.

Companies like Mojo Audio make high-performance LPS with multiple outputs that can be used to power multiple components simultaneously. By the time you spend $500 each on lesser performance LPS you could buy one high-performance LPS and power several components. Powering multiple components with one LPS will not only reduce the number of power cables and AC receptacles you need but will also eliminate potential ground loops.

#6 Cables:

Proper shielding from external influences, proper shielding between power and data wires, optimal conductivity, inductance, and capacitance, are all factors that can minimize corruption in your data stream.

Don’t confuse buffering, reclocking, and regenerating with error correction. Just because the data has been regenerated or reclocked doesn’t mean corrupted bits in the data stream have been corrected. Because few audiophile components correct corrupted data cables play a very important role.

Here’s a simple test to determine whether or not two components in your system have error correction: If they have error correction they will sound identical connected with a $20 cable or a $2,000 cable.

Don’t confuse price with performance. There are quite a few USB and Ethernet cables you can buy direct from the manufacturer for under $100 that sound better than big-name audiophile cables that sell for over $1,000. We did a comparison between some expensive audiophile Ethernet cables and a standard CAT8 Ethernet cable we paid under $10 for on Amazon. The $10 cable sounded notably better than audiophile Ethernet cables that sold for over 30 times the price. How is this possible?

Most big-name audiophile cable manufacturers have at least a 300% mark up on their cables. The reason for this is the retail stores want to be able to “build the sale” by convincing their customers that if they buy cables with their system they can offer them some amazing discounted price like 50% off. Of course even discounting the cables by 50% the store still makes a huge profit.

Next consider the cost of fancy connectors and coverings, fancy packaging, and full-page ads in audio magazines. With big-name audiophile cables most of what you're paying for have no effect on performance.

Before spending big bucks on cables from manufacturers who take out full-page ads in audio magazines I recommend comparing them with well-made commercial, pro audio, and direct-sale cable manufacturer's products. You may be quite surprised at what you hear.

#7 Player Software:

Most streaming services have their own free downloadable player software and most can be played through web browsers. These free players may give you access to all the features and recordings a streaming service offers, but they don’t necessarily deliver the highest sound quality.

High performance player software companies, such as Euphony Stylus, Roon, HQ Player, JRiver, and JPlay, have the ability to embed streaming services into their players to provide higher sound quality. Some even offer additional features and web radio in addition to playback from local music libraries. In addition to higher performance digital engines, these audiophile players often have advanced digital signal processing (DSP) such as file format conversion, upsampling, dithering, and digital EQ.

One of the reasons Roon is one of the most popular players is because of how well it integrates Tidal and Qobuz with web radio and playback from any storage on your network. By comparison HQ Player is nowhere near as user friendly as Roon and HQ Player can only integrate Qobuz (no Tidal). But HQ Player has one of the highest performance digital engines and offers advanced DSP such as dithering and upsampling.

Our favorite player software in recent years has been Euphony Stylus. Not only does Euphony come integrated with an optimized and minimized RAM-root Linux operating system, it comes integrated with Roon, HQ Player, Tidal, Qobuz, and YouTube. That means Euphony has tested a version of those various players and you can be assured of no glitches or conflicts. And each of those players sound notably better playing through Euphony's optimized Linux than through any operating systems we've heard.

Since we've started recommending Euphony to our customers we've not had a single issue or conflict: rock solid consistent performance. And 100% of our customers tell us that they prefer the sound of Euphony by itself than with Roon or HQ Player embedded into it. Sure, you get more features when you embed Roon or HQ Player, but the sound quality of Euphony by itself has led 100% of our customers to drop their former favorites and just use Euphony. I think that says it all.

Of course you mileage may vary. Since most of these players offer free trial versions I recommend comparing them side-by-side on your hardware. But remember that it is not just the specific player that determines the sound quality, but also the way you set up that player software.

#8 Software Settings:

Settings in your software can limit your resolution, convert music formats, or automatically upsample. Make certain to review all the settings for your player software and operating system. Often these settings are hidden or duplicated between player software and operating systems so it may require a bit of research.

One thing that will reduce sound quality would be if you incorrectly use upsampling. Only upsample to even multiples of the original recording. When you upsample to an odd multiple it requires interpolation which degrades sound quality. So upsample 44.1Khz to 88.2 or 176.4 or 352.8Khz not to 96, 192, or 384Khz.

There are other settings which may appear to be counter intuitive in some situations. For example, streaming HD files theoretically should sound better, right? Not always. Many songs that are so-called HD on streaming services are poorly upsampled 16/44.1Khz recordings. And many HD versions of songs don’t sound as good as the original 16/44.1Khz version. And if you have bandwidth issues with your ISP or in your LAN a lower resolution 16/44.1Khz recording may have better musical flow or emotional expression than an HD version of the same song.

The HD version of a song can require over 6X the bandwidth of the 16/44.1Khz version. That is no small difference. And since Delta-Sigma DACs use statistical error correction they would benefit from insane amounts of upsampling whereas R-2R DACs would only benefit from modest upsampling.

Then there's a matter of the optimal file format: Delta-Sigma DACs are native to DSD whereas R-2R DACs are native to PCM. Each type of DAC will sound better decoding their native file format.

These factors make the benefits of HD streaming, upsampling, and specific file formats, system dependent.

#9 Switches vs Filters:

There are components which go between your modem and your streamer that can improve sonic performance. Some are passive filters, like the Network Acoustics ENO and Muon that remove noise and DC offset from the data stream. Others are active components that buffer and regenerate the data.

Often people misunderstand what each component actually does to improve sound quality. Neither of these devices error correct so neither of these devices are able to repair data corruption. What both do is clean up the waveform to minimize the potential for data to be misread by the next component. A filter cannot improve the definition of the digital waveform. All it does is removes things that are unwanted. Routers and switches buffer and regenerate the waveform cleaning up and restructuring the data with better definition. They will misread noise and corrupted data and regenerate a well-formed but incorrect data stream.

Routers and switches can correct things that than passive filters can't. And passive filters can correct things that active devices can't. Depending on the specific type of data corruption and noise in your data stream you may benefit more from one, the other, or both of these devices.

The function of a router or switch is to direct the signal to multiple end points. They have one input and more than one output. A common misconception is that audiophile routers and switches improve system performance. That would depend on the specific router or switch and the specific components preceding or following it. If the preceding or following components are higher performance the router or switch would degrade rather than improve performance.

Unless you have audio systems in multiple rooms you might want to consider a device that has one input and one output such as an Ethernet filter or reclocker. During the process of buffering and regenerating the data the clocking is also regenerated. Any component that buffers data also regenerates and reclocks it. Just because something reclocks doesn't mean it improves the clocking.

#10 Reclocking:

The clocking frequencies used in computer and networking components are not optimized to the 44.1Khz and 48Khz multiples that are used in digital music. This is one reason audiophile switches and reclockers will improve sound quality. Another reason is reduced clocking noise.

Despite what some manufacturers may claim, you can’t hear the difference between the .005% accurate clocks used in consumer grade components and the .000005% accurate clocks used in audiophile components. What you can hear is the different types and levels of noise different clocks generate.

All clocks create hash and radiant noise. You may not be able to hear the difference between .005% and .000005% accuracy, but you can hear when hash noise is reduced to inaudible levels and radiant noise is shielded by the oven in an oven-controlled crystal oscillator (OCXO).

Many high-end audio manufacturers now offer high-performance OCXO clocks with their Ethernet, USB, I2S, and optical components. In addition to components with high-performance OCXO clocks, there are several companies making computer cards for USB, Ethernet, I2S, S/PDIF, and optical connectivity with the same audiophile clocking. These cards allow you to integrate this same level of performance into a modular streamer that will rival the best-of-the-best audiophile components.

Check out our blog on "Music Streaming without a Rat's Net" to show you how to integrate routers, switches, reclockers, isolators, master clocks, and power supplies, all into one elegant chassis. Would you rather have several components networked together with a rat’s nest of data and power cables, or just one component between your modem and your DAC?

Granted, having a multi-component multi-power supply networked system gives you easy plug-and-play upgrade options. But what do you think would sound better: a bunch of different components going though a rat's nest of cables, connectors, output transmitters, and input receivers. Or a modular single-chassis streamer where everything is directly connected through the main data buss of a computer?

Mojo Audio has several DIY and turn-key options to meet the needs of audiophiles of any budget or level of technical expertise. Versatile, upgradable, and obsolete proof. Something to consider.

A One Chassis Solution:

Rember where I mentioned the difference between multi-chassis and one chassis solutions?

Mojo Audio has several DIY and turn-key options for you.

The "One Box Wonder" is not an appliance, it's a modular system.

Our new Illuminati power supply module is engineered to utilize the existing heat sinks on the Streacom FC10 fanless chassis. With five independent linear power supplies you can isolate the power to your motherboard, SSDs, high-performance Ethernet cards, high-performance USB cards, and master clock.

Choose from the best-of-the-best cards, clocks, and modules for USB, Ethernet, I2S, AES/EBU, or optical, from companies like JCAT, SOtM, Sonare, M2 Tech, Pink Faun, After Dark, RME, and Lynx.

And use any Windows or Linux compatible software from companies like Roon, HQ Player, JRiver, Euphony Stylus, Audiophile Linux, JPlay, Tidal, Qobuz, and more.

Elegant. Versatile. Obsolete proof.

Hear it for yourself

Mojo Audio’s Mystique DACs have the purest digital conversion possible.

Our ultra-purist approach gives our Mystique DACs the organic character for which they are so famous. Tone and timbre that rivals the best of analog. Effortless micro-dynamics and incredible micro-details reveal previously unheard harmonics and spatial cues. A sense of breath and flesh that bring your music to life.

And with Mojo Audio’s 45-day no-risk audition, you can hear the Mystique for yourself, in your own system.

Want to learn more?

If you like what you've read check out my other blogs.

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Enjoy!

Benjamin Zwickel
Owner, Mojo Audio

7 Myths of Digital Audio Dispelled

Wed, 31 May 2023 13:52:46 +0000

UPDATED: 3.7.26

Introduction:

Everything in high-end audio is a compromise. No typology is best at everything. Every typology has its advantages and disadvantages. Digital is no different. Despite popular belief, there is not one type of DAC, quantization method, or CODEC, which is ideal for all situations, or better in all ways than others. Each has their advantages and disadvantages, and certain ones work better in certain situations.

Over the years I have heard and read all sorts of myths and misconceptions regarding digital audio. I hope to dispel the 7 most common categories of myths. I’m going to attempt to explain each typology in layman’s terms so that all of you can decide for yourselves what compromises you want to make.

Myth #1 Clocking:

Despite what you may hear in some company’s marketing message or read on forums; the accuracy of the clock makes no difference. The cheapest quartz clock they use in a $100 DVD player has more accuracy than you can hear, something like .005%. And you cannot hear the difference between .005% and .000005%.

So, what makes one clock sound better than another? Two things: how low the hash noise is that the clock creates in the audible spectrum and how little electromagnetic interference it creates. Both of these pollute other components in the same chassis.

An OCXO or Oven Controlled Crystal Oscillator, may have lower accuracy than a TCXO, or Temperature Controlled Crystal Oscillator, but it will sound better because it has lower hash noise. And the oven surrounding the clock does a better job of shielding electromagnetic radiation. BTW, OCXOs were invented to use in extremely cold environments, such as deep sea, arctic, or aerospace. None were created for high-end audio. It just so happens some of the best sounding clocks in high-end audio are oven-controlled crystal oscillators.

Here's a photo of the 10MHz OCXO clock (silver rectangle upper right) used in the new Jay's CDT-3 MkIII CD transport which is my current reference digital audio source. It is not the accuracy of the clock that makes it sound so good, but rather that it minimizes hash noise and electromagnetic radiation, so it does not negatively effect the surrounding circuitry. Notice this clock has a dedicated power supply and does not share power with any other devices in the chassis. The power supply isolation and proximity isolation of this clock, and how low the hash noise and electromagnetic interference is in the 10MHz clock they use, are what makes this transport sound so amazing, not how accurate the clock is.

So, what about Master Clocks? Think about it: they are isolated in a separate chassis with a separate power supply. That is what makes them sound so good, not that they are more accurate than integrated clocks.

Myth #2: DSD Sounds Better Than PCM or PCM Sounds Better Than DSD.

Does DSD, PCM , or MQA sound better than the others: the answer is not universal. Delta-Sigma DACs read native DSD and R-2R DACs read native PCM, so DSD will sound better on a Delta-Sigma DAC and PCM will sound better on an R-2R DAC. MQA is a CODEC not a quantization typology. I’ll get into CODECs a bit later.

Many of you are thinking “my DAC decodes all of those” which is another misconception. To begin with, Delta-Sigma DACs don’t “decode” they “interpolate” but I’ll get into that a bit later. So instead of the word “decode” let’s use the word “convert” which is what all digital to analog converters actually do.

Delta-Sigma DACs must first convert the PCM to DSD and R-2R DACs must first covert DSD to PCM before they can convert them to analog. When people make misguided universal statements that DSD is better than PCM or PCM is better than DSD they are correct only in respect to specific DACs.

Listening tests were performed and they concluded that there was no significant percentage of people who could distinguish the same recording quantized in PCM vs DSD. Of course, they did those tests 100% fairly using one of those rare DACs which has both an R-2R and a Delta-Sigma DAC chip. Since both chips used identical input stages, identical output stages, and identical power supplies. And since the chips were warmed up identically because they were in the same chassis. Those comparisons were fair. Very few other comparisons you may read about were done fairly.

Myth #3: Certain CODECs Sound Better

Many people confuse CODECs with quantization typologies. DSD, PCM and Wide-DSD are examples of quantization typologies. FLAC, ALAC, and MP3 are examples of CODECs. I as mentioned, MQA is a very advanced CODEC.

CODEC strands for “compression/decompression” and is a way to make audio files take up less storage space. This was a big deal a decade ago when storage space was expensive and large. Today most people can fit their entire music library on one SSD. Today compression is only relevant in terms of HD streaming bandwidth.

Some CODECs are what is called “lossless” which means they claim to decompress to exactly the original audio file. Even if they do decompress to the original audio file the processing required to decompress them lowers the performance of the player software they are being played on. For optimal performance you don’t want to store compressed files.

Interestingly enough, there are versions of FLAC and ALAC which are not compressed. So though called the same thing as the compressed versions, they are most certainly not CODECs. What these formats do which is desirable is package the album cover art and advanced metadata with the music data.

WAV is the most basic form of music file, and is not a CODEC. WAV is what you find on a CD. WAV can have basic metadata with the album and track names but cannot have album cover art packaged with the music data. Of course, player software can associate the album cover art with a specific WAV file yielding the same convenience of uncompressed FLAC and ALAC, with better performance.

WAV sounds the best because it is the least complex and requires the least amount of processing. And WAV is universally played on most player software. FLAC is optimized for Windows and ALAC is optimized for Apple, so each tends to work and sound best when played through their respective operating systems. Since Apple is a fancy GUI on top of Unix, ALAC will tend to sound better in Linux.

Of course, the specific player software you are using can have more of an impact on performance than the file format. So, you may want to compare different file formats or CODECs on your specific player software before deciding what will sound best for you. My guess would be that WAV will sound best on all, but that may not necessarily be true.

Myth #4: Delta-Sigma Sounds Better Than R-2R or R-2R Sounds Better Than Delta Sigma.

There are basically two quantization typologies commercial music is marketed in: PCM and DSD. There is Wide-DSD, aka 5-bit or 8-bit DSD. Though widely used in studios during the recording, mixing, and mastering process, sadly there are no commercially available versions of our favorite recordings in Wide-DSD, even though a significant percentage of Delta-Sigma DAC chips can convert it.

All DAC chips, discrete or integrated, use R-2R and/or Delta-Sigma conversion. Some are “Hybrid” DAC chips which use both R-2R and Delta-Sigma. For the most significant bits they use R-2R, for the least significant bits they use Delta-Sigma, and they use an algorithm to weight and combine the two.

This same weighting and combining can also be done with more than one R-2R resistor ladder. This concept is called “segmented conversion.” The famous 24-bit PCM1704 DAC chip, often mistakenly called R-2R, is an excellent example of an IC segmented R-2R DAC chip. The greatest bit depth you can laser match on one IC’s Silicon wafer is 20-bits. Something like .000005% matching tolerance is required to achieve a bit depth of 20-bits. To get 24-bits of digital resolution (note I said “digital resolution”) you need more than one resistor ladder and an algorithm to weight and combine them.

Those so-called discrete R-2R DACs are actually segmented R-2R. They use modest tolerance resistors commercially available, such as .5%, put them in multiple lower bit resistor ladders, weight the voltage coming out of each resistor ladder, and then use algorithms to combine the weighted voltages from each ladder. This is one reason why segmented R-2Rs made from discrete parts have relatively poor linearity. Discrete R-2R DACs are a brilliant piece of engineering and can have a very attractive sound. To me they sound halfway between a true R-2R and a Delta-Sigma DAC. Not my cup of tea, but for some of you this may be the best of both worlds.

In any event, R-2R DACs, segmented or single ladder, are the only DAC typology which decode what’s in the bit stream. They take each digital word, which could be 16-bits or 24-bits, they put it through a series to parallel shift register, and then each bit is put through a rung in a resistor ladder, and then all the voltages from all the bits are summed in a summing amplifier.

For those of you who don’t completely understand digital theory let me get back to basics. In binary digital quantization each bit has double the value of the previous. 1, 2, 4, 8, 16, 32, 64, 128, and so on. Then you combine these numbers to create any number: 1 + 2 + 4 = 7 or 32 + 128 = 160. These numbers correspond to relative voltages or volumes, aka quantization values, in a PCM digital recording.

One thing many people find frustrating is the better your DAC gets, the more flaws you’re going to hear in your recordings. With R-2R you hear what is in the recording, warts and all. I prefer the articulation, proper time and tune, and harmonic coherency of R-2R. But I can’t tell you how many times I’ve played a modern recording and found it fatiguing. With R-2R, as your DAC gets better, your best recordings will sound transcendent while your worst recordings will become unlistenable.

As I mentioned, Delta-Sigma DACs, which comprise over 95% of the DAC chips sold today, do not actually “decode” the bit stream but rather "interpolate" it. They take in the digital bit stream faster than the music is playing, analyze it, noise shape it, error correct it, interpolate what they think the musical signal was supposed to look like, and then output a flawless waveform. Not quite the waveform which was quantized, but a very smooth and very even waveform. That is why Delta-Sigma DACs sound so smooth and refined. This is also why Delta-Sigma DACs have an advantage when playing mediocre sources such as music streamed from the internet.

Of course, algorithms cannot tell the difference between a bit-read error and emotional content. Think about it: emotional content is when the musician plays harder/softer or faster/slower. The algorithms in Delta-Sigma DACs see those as different than the other times similar notes were played, and they attempt to correct something that does not need correcting. I jokingly call statistical error correction a “defunkification filter.” Not my cup of tea, but I can see why so many people love that super smooth hear-no-evil sound. Sometimes a bit of tasteful color hides many sins.

FPGA DACs are the heavy weights in the Delta-Sigma world. They have much more powerful processors, so they can run much more sophisticated algorithms than a normal Delta-Sigma DAC. We are talking about super statistical error correction combined with insane upsampling. And the manufacturer can evolve and improve their algorithms and update their firmware. So, FPGA DACs are the most obsolete proof of all the DAC typologies. If you like that smooth, refined sound, and you’re planning on keeping your DAC for years to come, you’ll probably love an FPGA DAC.

Myth #5: The Higher the Upsampling, the Better the Sound

When some people claim upsampling sounds better and others claim it sounds worse: they are both correct. If their DAC uses statistical error correction, they are correct that insane upsampling can sound better and if their DAC does not use statistical error correction, they are correct that insane upsampling can sound worse. All this stuff about higher resolution sounding better is not universal, but DAC dependent.

44.1KHz done right sounds amazing. But it does have digital artifacts in the audible spectrum at fractions of 44.1KHz, such as 22.05KHz and 11.025KHz. By upsampling from 44.1KHz to 88.2KHz or 176.4KHz you move those digital artifacts above the audible spectrum and you get a subtle improvement in smoothness and a bit more harmonic coherency.

Upsampling can be done by some CD transports or can be done on a computer music server using Player software. Some player software will do upsampling and others will not. HQ Player is considered by many to be the heavy weight in the upsampling player software world.

As a rule, an R-2R DAC would not have statistical error correction. But theoretically you could add statistical error correction to an R-2R DAC if you feed it from a field programmable gate array (FPGA) running the right algorithms. An FPGA is another place you can do upsampling. Most FPGAs I’ve heard doing upsampling sounded better.

As a rule I've loved upsampling on a wide range of CD transports. Generally, I never loved it on my computer audio. Of course I only listen to R-2R DACs, so that would make perfect sense. Upsampling is very component and system dependent and over the years customers have reported both positively and negatively regarding upsampling.

Myth #6: Higher Resolution Files Sound Better

Despite what you might hear in some company’s marketing message or might read on a forum, Dr. Nyquist knew what he was talking about, and sampling at 44.1KHz perfectly quantizes the audio wave in frequencies up to 22KHz. My current reference digital source is a 16-bit 44.1KHz Red Book only CD transport. It sounds better than any high-definition 24-bit 352.8KHz PCM file, SACD, DSD, MQA, or anything else you can name played on computer audio.

Without upsampling 16-bit 44.1KHz Red Book CDs, played on a proper CD transport, will sound better than any HD computer audio I’ve ever heard. Granted I've not heard everything. But I've heard some of the best-of-the-best in computer audio, and there is no comparison, in terms of time, tune, tone, and timbre, harmonic coherency, and attack - bloom - decay, to a proper CD transport. Seriously: no comparison. The best CD transports use the top loading clamped disk CD typology and one of the vintage Red Book CD only mechanisms. Note that all things being equal, CD transports which use multi-disk DVD drives and tray loading CD transports can never sound as good as a simple Red Book 16-bit 44.1KHz only old school CD mechanism with a top clamp.

Myth #7: One transfer protocol sounds better than all others.

Transfer protocols are things like USB, Ethernet, S/PDIF, and I2S. Most computer transfer protocols, such as USB and Ethernet, are asynchronous, meaning they buffer and reclock the incoming data.

This is not different than what an external USB or Ethernet reclocker/regenerator does. So theoretically, if the internal USB or Ethernet input in a specific DAC is better than a specific external USB or Ethernet reclocker/regenerator, adding that “magic box” will degrade rather than improve performance.

S/PDIF stands for Sony/Philips Digital Interface and is one of the oldest and most commonly used digital transfer protocols. All CD transports and most DVD playes output S/PDIF coaxial with an RCA jack. S/PDIF is not inherently a computer transfer protocol, but there are some servers and streamers which output S/PDIF, and there are USB and Ethernet to S/PDIF converters.

AES is the balanced version of S/PDIF: same code with dual phase and better shielding. AES was created for pro audio to minimize noise when running longer distances. Most of the better CD transports output AES. In most cases I’ve found the coaxial S/PDIF can sound equal and even better than balanced AES over the normal 1-meter cables used in most home audio. This is partially because a good low-mass RCA connector is made from high-purity copper or silver whereas most balanced XLR connectors are made from less conductive high copper content brass. Same thing with BNC: the metallurgy of the connector can be a more important factor than being a perfect 75 ohms.

One of the big advantages of S/PDIF or AES is that they use an external clock. This lowers the potential noise inside the DAC’s chassis. Several highly regarded DAC manufacturers don’t integrate a USB or Ethernet input into their DACs, but rather make a high-performance USB or Ethernet to S/PDIF or AES converter, to better isolate noisy computer audio from their sensitive DAC.

Some DACs buffer and reclock the S/PDIF signal. This has advantages and disadvantages. When using an inferior source, such as a $1,000 multi-disc DVD player, the additional buffering and reclocking will improve performance. But when using a world-class CD transport with uncompromising clocking, the internal buffering and reclocking will degrade rather than improve performance.

I2S was engineered as an internal transfer protocol for inside of DACs and ADCs and is the language most DAC chips read. In most DACs all other transfer protocols are converted to I2S before they can be sent to the DAC chip. The official specification for I2S is that it should not be used for longer than 4”. This is why so few companies sell I2S compatible CD transports or DACs: it is not necessarily a good idea.

Think about it: all other transfer protocols are a bit stream with embedded clocking. Companies who boast about the performance of their I2S claim that the clocking in a single bit stream becomes corrupted. You see I2S has three wires: the data stream with embedded clocking, a bit clock which synchronizes with each bit, and a word clock which synchronizes with each digital word. If clocking in data streams can get corrupted, then why would it make sense to try to synchronize three data streams and clocks?

The only reason I2S sounds better on a specific DAC is because the other transfer protocols are of a lower level of performance. In a sense I2S saves the manufacturer money in that they are relying on expensive clocking from the component feeding their DAC rather than integrating such high-performance clocking.

So, which transfer protocol has the best sound? That would depend on the digital source (server, streamer, or CD transport), and the quality of the specific digital input on a specific DAC. Most DACs don’t have the same performance from all their inputs. Many DAC manufacturers will even state their best input is USB or Ethernet or S/PDIF. And even if you have the best input on your DAC, if you’re using a less than optimal digital source, overall performance won’t be all that good. So, once again, transfer protocols are not universal, but highly component dependent.

Conclusions:

So as you can see, different DAC typologies, different quantization formats, and different CODECs, each work best under specific situations. Different CODECs sound better with different operating systems and player software. And upsampling and different transfer protocols are both very hardware and system dependent.

If your main digital source is an HD streaming service, or a less than optimal computer or transport, you may very well prefer the smoothing sound of statistical error correction and Delta-Sigma DACs. Remember: FPGA DACs are the heavyweights in the Delta-Sigma world. If you are a purist and want the optimal time, tune, tone, timbre, and harmonic coherency, then you very well may prefer an R-2R DAC.

The most important thing is to be open minded and not assume anything. Always evaluate components in the system they will be played in using the digital source and software that will be feeding it. Even if you've found something like upsampling or a USB reclocker, or a specific CODEC sounded best with your current DAC or digital source, when considering new digital source components, do some blind A/B comparisons with friends, where the listener does not know what they are hearing. I've always found blind A/B comparisons are the best way to evaluate new components.

Hear it for yourself

Mojo Audio’s Mystique DACs have the purest digital conversion possible.

And with Mojo Audio’s 45-day no-risk audition, you can hear the Mystique for yourself, in your own system.

Want to learn more?

If you like what you've read check out my other blogs.

And sign up for our e-newsletter.

Enjoy!

Benjamin Zwickel
Owner, Mojo Audio

The 24-Bit Delusion

Fri, 26 May 2023 13:55:03 +0000

UPDATED: 3.7.26

Introduction:

More and more music has become available in “high-definition” (HD) digital formats, such as 24-bit 192KHz downloads, 24-bit 88.2KHz MQA streaming, and DSD. Now I hear talk about developing a new 32-bit 384KHz standard for HD music. Interestingly enough, not everyone agrees that greater bit depth and higher sampling rates are good things.

This blog will explain the math and physics of digital recording and musical reproduction in layman's terms so that you can decide for yourself if this is progress or simply marketing madness.

If you're not sure if you should believe the statements in this blog which contradict much of the marketing hype, myth, and legend in the audiophile industry, feel free to check the references at the end.

You also may want to refer to my other blog on “DSD vs. PCM: Myth vs. Truth.”

Bits, Bytes, and Digital Words:

So why did 24-bit become the new standard?

When digital data is transferred and manipulated it is moved in bytes rather than as individual bits. There are 8 bits to a byte and a byte is known as a digital word. This is why everything in the digital world is divisible by 8. So 16 bits = 2 bytes and 24 bits = 3 bytes. Both 16 bits and 24 bits became standards because each represented the next digital word.

Historical note: The 16-bit format existed long before 16-bit digital-to-analog converters (DACs) were available.

Sampling Rate and Bit Depth:

The process of converting analog sound waves into a digital format is known as “quantization,” which is often represented as points plotted on an XY axis. The horizontal X axis represents time or sampling frequency and the vertical Y axis represents amplitude or bit depth. In the graphic below the white wave form represents the musical signal being quantized and the green step pattern overlaid represents the quantized values.

Sampling rate is the frequency at which the amplitude of the analog sound wave is sampled. The 44.1KHz sampling frequency specified for Red Book CDs sample the amplitude of the music 44,100 times each second. The 96KHz sampling frequency used in the 7.1 channel audio embedded into DVDs and Blu-Rays sample the amplitude 96,000 times each second.

Bit depth translates to the number of steps the amplitude of the analog sound wave is divided into at each sampling. A 16-bit recording has 65,536 steps, a 20-bit recording has 1,048,576 steps, and a 24-bit recording has 16,777,216 steps. Yes, you read that correctly: a 24-bit recording has 256 times the number of amplitude steps as a 16-bit recording.

The more bits and/or the higher the sampling rate used in quantization, the higher the theoretical resolution. So a 16-bit 44.1KHz Red Book CD has 28,901,376 sampling points each second (44,100 x 65,536). And a 24-bit 192KHz recording has 32,212,254,000,000 sampling points each second (192,000 x 16,777,216). This means 24-bit 192KHz recordings have over 111,455 times the theoretical resolution of a 16-bit 44.1KHz recording. No small difference.

So why is it that HD recordings sound only slightly better than a 16-bit 44.1KHz recordings made from identical masters? Later in this blog I’ll explain the difference between theoretical and actual resolution.

Dynamic Range and Bit Depth:

Dynamic range is the difference in volume between the quietest and the loudest passage. Dynamic range is measured in decibels (dB).

Just for reference, here are some examples of dynamic range that most of us can relate to:

The sound of a mosquito flying 3 meters away is 0dB.
The hum of an incandescent bulb at 1 meter away is 10dB.
The background noise in a quiet recording studio is 20dB.
The background noise in a normal quiet room is about 30dB.
Early analog master tape had a dynamic range of only 60dB.
LP micro-groove records have a dynamic range of 65dB.
Dolby increased analog master tape dynamic range to 90dB.
The sound of a jackhammer at 1 meter away is 110dB.
The sound of a full orchestra at 1 meter away is 120dB.
Over 130dB causes irreparable hearing loss.
The sound of a jet aircraft at takeoff is 140dB.

In a digital recording 1-bit = 6dB:

16-bit Red Book CDs have a dynamic range of 96dB.
20-bit digital master tape has a dynamic range of 120dB.
24-bit HD formats have a dynamic range of 144dB.

But wait…isn’t the background noise in a quiet room 30dB?

So you can’t actually hear the difference between the dynamic range of a 16-bit recording and a 20-bit recording unless you turn the volume up high enough above the 30dB background noise that it would cause hearing damage.

So why on Earth would they even create a digital recording format that can't even be listened to?!?!?!?!?

Simple: bit-depths and sampling rates far above the range of human hearing are used during the recording, editing, mixing, and mastering processes to lower digital noise in audible spectrum when recordings are downsampled to the significantly lower resolution sold in commercially released recordings.

Noise Floor:

Dynamic range is the loudest possible sound and noise floor is the quietest.

We already know that a quiet room has a background noise level of about 30db that we need to rise above. Even if the system is playing above the 30db room noise, the power supply in a DAC will mask the LSB if the peak-to-peak voltage of the noise in the power supply is not less than the voltage of the LSB.

In order for a DAC to actually resolve a specific bit depth the peak-to-peak voltage of the ripple in the power supply has to be lower than the voltage of the LSB. And in order for a DAC to resolve a specific sampling rate the speed of the power supply has to be faster than the sampling frequency.

Based on a 2.5V output of a single-ended DAC (about average), below are the voltages power supply noise must be below in order to hear the LSB:

16-bit LSB noise floor voltage = 76uV
20-bit LSB noise floor voltage = 4.75uV
24-bit LSB noise floor voltage = 0.3uV

For a reference, the common LM317 power regulator, the quality used in most commercial electronics, has about 150uV peak-to-peak noise and the best ultralow-noise power regulators used in the best-of-the-best of audiophile electronics have about 5uV of peak-to-peak noise. So even the 5V output of a balanced DAC could not resolve anything close to the LSB voltage of a 24-bit recording.

Sorry to burst anyone's bubble and contradict the marketing hype, myth, and legend in the audiophile industry, but just because a DAC is capable of decoding 24-bits doesn't mean it is capable of actually resolving that bit-depth in its analog output stage.

According to the experts who manufacture the finest DAC chips, resistors, and power regulators, there is theoretically no way to make electronics that are capable of discerning much greater than a 20-bit resolution (120dB dynamic range). Any company who claims 24-bit resolution from their DAC is simply full of shit. Oh they can decode 24-bits, because 24-bits does exist on the digital side, but the analog output stage in the world's best DACs are not capable of resolving much more than 20-bits of dynamic range.

And don't even get me started on DACs with tube output stages: the lowest noise floor of a tube output stage is about 90dB which means despite whatever a manufacturer may claim no tube DAC can even resolve the dynamic range in a 16-bit recording let alone a 24-bit recording.

Theoretical vs. Actual Resolution:

According to Dr. Nyquist's theorem, sampling at twice the maximum audible frequency yields a perfect reproduction of the audio waveform. Any higher resolution will only plot more points along the same curves.

So in order to correctly sample a 20KHz note, the maximum frequency human ears can hear, you would need to sample at greater than 40KHz. The 44.1KHz sampling rate of a Red Book CD was engineered to allow a 20KHz sound to be recorded accurately.

Sorry to be the one to burst your bubble, but despite what many audiophiles may believe, less than one person in a thousand can hear anything above 20KHz as a child and there is almost no one over the age of 40 who can hear much above 15KHz.

So why would there be any need for higher sampling frequencies than 44.1KHz?

One reason is quantization noise. Since quantization noise is present around the sampling frequency of a PCM recording, a 44.1KHz recording has quantization noise one octave above the human hearing limit of 20KHz. This quantization noise needs to be filtered out, so all DACs have a low-pass filter at the output. Because the quantization noise is only one octave above audibility the filters used have a very steep slope so as to not filter out desirable high frequencies. These steeply sloped low-pass digital filters are commonly known as "brick wall" filters.

Though you hear a lot about "brick wall" filters causing an audible distortion in the top end of early Red Book CD players , the fact is that was only a small part of the reason early Red Book CDs and players had an unnatural sounding top end. Most of the hard, harsh, unnatural sounding high frequencies in early digital had more to do with flaws in the power supplies and flaws in the recording process, not "brick wall" filters.

In order to lower the quantization noise in the audible spectrum professional formats, such as 24-bit 352.8KHz DXD, were developed for recording studios. The reasons 24-bit DAC chips were developed was so recording engineers could monitor their their recording, editing, mixing, and mastering in real-time without having to downsample. Of course the companies who produced DAC chips stopped producing the lower resolution DAC chips. And companies who manufactured consumer electronics used these 24-bit DAC chips and began to make creative marketing claims about their products.

Even though many recordings are advertised as being 24-bit, only a small portion of the 24 bits of dynamic range are actually used. These so-called 24-bit recordings are compressed down to a dynamic range that most electronics are capable of producing. I'm not talking high-end audiophile electronics, but rather your average car stereo, phone, or MP3 player. Commercial recordings with more than 40dB of dynamic range have peaks which would clip out most electronics at a very low volume. There are more details on how dynamic range effects electronics in the following section on "Playback Equipment Requirements."

So what do they do with commercially marketed so-called 24-bit recordings? They simply fill in the Most Significant Bits (MSB) with 1s and the Least Significant Bits (LSB) with 0s and center the actual dynamic range. Even most of the best of audiophile recordings have less than 70dB of dynamic range. They could have released a recording of identical performing in 16-bits, but because naive consumers have been tricked into believing the BS marketing messages regarding 24-bits, the record companies put an average of 5-7 bits of dynamic range in a 24-bit format. How silly.

DSD is no different. Though you can't directly relate DSD in terms of bit depth and sampling frequency, a rough estimate is that DSD64 (aka SACD or single-rate DSD) is fairly close in resolution to a 24-bit 88.2KHz PCM recording. But instead of having quantization noise centered around the sampling frequency like PCM, DSD64 has significant amounts of digital noise just above 25KHz, as is shown in the graphic below.

To get around this problem Delta-Sigma DACs have noise-shaping algorithms and many upsample to higher frequencies to move the quantization noise to a high enough frequency so that it can be filtered out with a minimum of distortion in the audible range. This is one of the reasons why computer audio player software that upsamples DSD64 to Double-Rate or Quad-Rate DSD makes such an improvement in Delta-Sigma DAC performance. This is also one of the reasons why upsampling to high rates improves the performance of PCM files decoded by most Delta-Sigma DACs.

Another reason why upsampling improves the performance in Delta-Sigma DACs is that they use statistical error correction algorithms, so the more data points, the more accurate the error correction. This is what tricks many audiophiles into believing that higher sampling frequencies above the 2X bandwidth that Dr. Nyquist stated will yield higher resolution. This is not true for R-2R ladder DACs. Upsampling to 88.2KHz is enough to remove any digital artifacts from the audible spectrum when using an R-2R DAC.

For more detailed information on this topic refer to my blog on “DSD vs. PCM: Myth vs. Truth.”

Another consideration of higher sampling rates and greater bit depth is system resources. Both require more storage space, more RAM, faster processors, and higher current power supplies. Though the optimal sampling frequency and bit-depth which are required to reproduce accurate music are a matter of heated debate, there is no doubt that excessive resolution unnecessarily uses up system resources and unnecessarily increases the size and cost of components. Also note that the faster processors and higher current power supplies required to process higher resolutions or to do high upsampling are inherently noisier.

Playback Equipment Requirements:

There are very few systems, even among the best-of-the-best, which can accurately play back the full 120dB dynamic range of a 20-bit recording. This is why few commercial recordings have higher than 6-bits of dynamic range , let alone the 144dB dynamic range of a 24-bit recording. Keep in mind that the maximum dynamic range of micro-groove LP records is only 65dB and the maximum dynamic range of pro-audio Dolby analog master tapes is 90dB.

There was wisdom to the LP record and the analog tape standards. The relationship between amplifier wattage and decibels (dB) of volume is logarithmic not linear. Manufacturers knew that for every 3dB consumers raised their volume, they would have to double the wattage of their amplifier and double the output of the speakers. So keeping the dynamic range of consumer recording under 60dB is much of what allowed home entertainment equipment to be affordable, of modest size, and relatively high-fidelity.

So that 120dB live music can be played on most systems, recording studios limit the dynamic range using a process called “dynamic compression.” The process of dynamic compression makes the quieter passages relatively louder and the louder passages relatively more quiet. This makes it easier to discern low-level details from the louder passages. When music is properly dynamically compressed it allows you to listen at a reasonable volume and still hear all the subtle harmonic cues that reveal the tone, timbre, and room acoustics in a recording.

Think about it: a 60dB dynamic range on top of a 30dB background noise equals 90db. How much louder than 90dB do you want to listen to music in your home? More importantly, for every additional 3dB you increase dynamic range, you would need to double the wattage of your amplifier and double the output of your speakers.

All things being equal, to go from 90dB output up to 99dB, you would need an amplifier with 8 times the wattage and speakers with 8 times the output. To accurately reproduce a recording at 120dB, you would need an amplifier with 1,024 times the wattage and speakers with 1,024 times the output than it would take to reproduce the same recording at 90dB. I don’t know about you, but a system like that will neither fit in my room nor my budget.

Summary:

Well, all that’s a real ear opener, isn’t it?

When people claim to hear significant differences between 16-bit and 24-bit recordings it is not the difference between the bit depths that they are hearing, but most often the difference in the quality of the digital remastering. And most recordings are engineered to sound best on a car stereo or portable device as opposed to on a high-end audiophile system. It’s a well-known fact that artists and producers will often listen to tracks on an MP3 player or car stereo before approving the final mix.

The quality of the recording plays a far more significant role than the format or resolution it is distributed in. But to increase profits, many modern recording studio executives insist that errors be edited out in post-production, significantly compromising the quality of the original master tapes. So no matter what format these recordings are released in, the music will always sound mediocre, since you can never have higher performance than what is on the original masters.

In contrast, some of my favorite digital recordings were digitally mastered from 1950s analog recordings. Many of these recordings were done as a group of musicians playing in a room with one take per track and no post-production editing. Though these recordings have much higher background noise being limited by old-school pre-Dolby 60dB dynamic range master tape, they retain an organic character and in-the-room harmonic cues that can't be duplicated any other way.

Hear It for Yourself:

Mojo Audio’s Mystique DACs have the purest digital conversion possible.

And with Mojo Audio’s 45-day no-risk audition, you can hear the Mystique for yourself, in your own system.

Want to learn more?

If you like what you've read check out my other blogs.

And sign up for our e-newsletter.

Enjoy!

Benjamin Zwickel
Owner, Mojo Audio

References:

http://www.lavryengineering.com/lavry-white-papers/

http://www.highendnews.info/technology/oversampling_and_bitstream_metho.htm

http://www.grimmaudio.com/site/assets/files/1088/dsd_myth.pdf

http://bitperfectsound.blogspot.com/2014/12/dst-compression.html

http://www.soundonsound.com/sos/sep07/articles/digitalmyths.htm

http://www.digitalpreservation.gov/formats/fdd/fdd000230.shtml

https://en.wikipedia.org/wiki/Direct_Stream_Digital

http://hometheaterreview.com/super-audio-compact-disc-sacd/

http://en.antelopeaudio.com/blog/

http://benchmarkmedia.com/blogs/news/15121729-audio-myth-24-bit-audio-has-more-resolution-than-16-bit-audio

Note: many of the graphics used in this blog were adapted from graphics taken from these reference sources.

DSD vs. PCM: Myth vs. Truth

Fri, 26 May 2023 13:52:09 +0000

UPDATED: 3.7.26

Introduction:

Direct Stream Digital (DSD) has become a big thing in high-end digital audio. Simplified encoding and decoding, along with ultra-high sampling frequencies, promise unparalleled performance. Is this what we’ve all been waiting for or just mass-marketing hype? This blog separates the hype from the technical facts. I’ll explain in what ways DSD has the advantage and in what ways pulse-code modulation (PCM) is better.

You also may want to refer to my other blog on “The 24-Bit Delusion.”

A Brief History:

In 1857, Édouard-Léon Scott de Martinville invented the phonautograph, which could graphically record sound waves. In early 1877, Charles Cros devised a way to reverse that process on a photoengraving to form a groove which could be traced by a stylus, causing vibrations that could be passed on to a diaphragm, recreating sound waves.

In late 1877, Thomas Edison used Cros’ theories to invent the cylinder phonograph, allowing music lovers to experience recorded music in their homes for the first time. Can you imagine a modern cylinder phonograph? Tangential tracking…no arc error…no skating error. The concept was flawless.

In 1887, Emile Berliner invented the technically inferior disk phonograph. Disks warp and there was arch error and skating errors introduced. Certainly no comparison to the tangential tracking Edison cylinder player.

But since disks are much cheaper to produce than cylinders, and since disk fit nicely in display bins at stores and can include larger cover art and notes, they became the standard. And so began the long history of the recorded music industry being more about consumer convenience and optimal profits than about optimal fidelity.

The digital revolution was no different. Philips and Sony collaborated on the new standard for a consumer digital format in 1979. Philips wanted a 20 cm disk, but Sony insisted on a 12 cm disk which could be played in a smaller portable device. In 1980 they published the Red Book CD-DA standard, and mass-market digital music was born. Many in the recording industry in the early days of digital joked that CD stood for “compromised disk.”

In the early 1980s, when digital recording became readily available, studios converted from analog to digital to save money. For studios, this cost less for the equipment, required less space for both recording and archiving, and made it easier to mix and edit tracks in post-production. For consumers, there weren't many advantages. Most of the early digital recordings were produced with relatively low resolution and sounded so fatiguing they would make you want to tear your ears off.

The switch from PCM to DSD was no different. In the early 1990s Sony wanted a future-proof, less expensive medium to archive their analog masters. In 1995 they concluded that storing a 1-bit signal directly from analog-to-digital would allow them to output to any conceivable consumer digital format (LOL...later I'll explain how Sony screwed the pooch on this decision). This new 1-bit technology was achieved by outputting from the monitoring pin on Crystal’s new 1-bit 2.8Mhz Bit Stream DAC chip.

Later, Sony’s consumer division caught wind of DSD and collaborated with Philips to create the SACD format. Of course, from the time the SACD was conceived until the time it came to market, DAC chip manufacturers had advanced from 64fs to a higher 128fs sampling rate (aka Double-Rate DSD) and from 1-bit to a higher-resolution 5-bit wide-DSD format. If the SACD format was DSD128 instead of DSD64 and 5-bits instead of 1-bit it would have made a huge difference in performance. Oops.

Long before the DVD, SACD, or DSD formats were developed, the Bit Stream DAC chip was introduced to the consumer market as a lower-cost alternative to the significantly more expensive R-2R multi-bit DAC chip. Bit Stream DAC chips have built-in algorithms to convert PCM input to DSD, which is then converted to analog. Once again, the result was a huge cost saving at the expense of fidelity.

It was in part Bit Stream DAC technology which allowed the development of our modern 7.1 channel audio that’s embedded into video formats. This also allowed electronics manufacturers to market DVD players in small chassis with cheap power supplies which could retail for under $70. Once again, the audio purist never stood a chance.

In contrast, not only do multi-bit R-2R DAC chips cost significantly more to manufacture than single-bit DAC chips, but they also require much larger and more sophisticated power supplies. If you were to make a 7.1 channel R-2R multi-disk player, it would cost several times the price of Bit Stream technology and it would be several times the size. Certainly not what the average consumer is looking for.

To sum things up, the recorded music industry has made decision after decision to maximize profits and mass consumer appeal at the expense of the audio purist. History lesson over.

DSD vs. PCM Technology:

PCM recordings are commercially available in 16-bit or 24-bit and in several sampling rates from 44.1KHz up to 192KHz. The most common format is the Red Book CD with 16-bits sampled at 44.1KHz. DSD recordings are commercially available in 1-bit with a sample rate of 2.8224MHz. This format is used for SACD and is also known as DSD64 or single-rate DSD.

There are more modern, higher-resolution 1-bit DSD formats, such as DSD128, DSD256, and DSD512 as well as wide-DSD formats with 5-bit to 8-bit Delta-Sigma decoding which I will explain later. These formats were created for recording studios and comprise only a very small portion of the recordings which are commercially available.

Though you can’t make a direct comparison between the resolution of DSD and PCM, various experts have tried. One estimate is that a 1-bit 2.8224MHz DSD64 SACD has similar resolution to a 20-bit 96KHz PCM. Another estimate is that a 1-bit 2.8224MHz DSD64 SACD is equal to 20-bit 141.12KHz PCM or 24-bit 117.6KHz PCM.

In other words a DSD64 SACD has much higher resolution than a 16-bit 44.1KHz Red Book CD, roughly the same resolution as 24-bit 88.2KHz PCM recording, and not as much resolution as a 24-bit 176.4KHz PCM recording.

Both DSD and PCM are “quantized,” meaning numeric values are set to approximate the analog signal. Both DSD and PCM have quantization errors. Both DSD and PCM have linearity errors. And both DSD and PCM have quantization noise that requires filtering at the output stage. In other words, neither one is perfect.

PCM encodes the amplitude of the analog signal sampled at uniform intervals (sort of like graph paper), and each sample is quantized to the nearest value within a range of digital steps. The range of steps is based on the bit depth of the recording. A 16-bit recording has 65,536 steps, a 20-bit recording has 1,048,576 steps, and a 24-bit recording has 16,777,216 steps.

DSD encodes music using pulse-density modulation, a sequence of single-bit values at a sampling rate of 2.8224MHz. This translates to 64 times the Red Book CD sampling rate of 44.1KHz, but at only one 32,768th of its 16-bit resolution.

In the above graphical representation of PCM as a dual axis quantization, and DSD as a single axis quantization, you can see why the accuracy of DSD reproduction is so much more dependent on the accuracy of the clock than PCM. Of course, the accuracy of the voltage of each bit is just as important in DSD as PCM, so the regulation of the reference voltage is equally important in both types of converters.

Of course the accuracy of the clocking during the recording process which is done at several times the resolution of commercial DSD64 SACD and 16-bit 44.1KHz PCM recordings is significantly more important than the accuracy of the clocking of either DSD or PCM during playback.

There are other DSD formats which use higher sampling rates, such as DSD128 (aka Double-Rate DSD), with a sampling rate of 5.6448MHz; DSD256 (aka Quad-Rate DSD), with a sampling rate of 11.2896MHz; and DSD512 (aka Octuple-Rate DSD), with a sampling rate of 22.5792MHz. And most modern A to D and D to A Delta-Sigma converters do multibit wide-DSD with 5-bits to 8-bits decoding in parallel. All of these higher-resolution DSD formats were intended for studio use as opposed to consumer use, though there are some obscure companies selling recordings in these formats.

Note that Double, Quad, and Octuple DSD have both the potential for a 44.1KHz multiple and a 48KHz multiple sample rate for 100% equal division down to DSD64 SACD and 44.1KHz Red Book (both 44.1KHz multiples) or 96KHz and 192KHz High-Definition PCM formats (both 48KHz multiples).

Of course when studios convert a 48KHz multiple format to a 44.1KHz multiple format or visa versa they introduce quantization errors. Sadly this is often the case with older recordings when they are released in a remastered 24-bit 192KHz HD version derived from DSD64 masters, such as the ones Sony and other companies used to archive their analog masters in the mid-90's. Note that the optimal HD PCM format which can be created from a DSD64 master would be 24-bit 88.2KHz. Any sampling rate over 88.2KHz or that is equally divisible by 48KHz would have to be interpolated (not good). But consumers demand 24-bit 192KHz versions of all their old favorites, so companies provide them, despite the known consequences.

The Problems:

There are three major areas where both PCM and DSD fall short of perfection: quantization errors, quantization noise, and non-linearity.

Quantization errors can occur in several ways. One way which was most common in the early days of digital recording had to do with the resolution being too low. Think of the intersection points on a piece of graph paper. You can’t quantize to a fraction of a bit, and you can’t quantize to a fraction of a sampling rate. You can only quantize to a value which falls on the intersection points of bit-depth and sampling rate. When the value of the analog signal falls between two quantization values, the digital recording ends up recreating the sound lower or higher in volume and/or slower or faster in frequency, distorting the time, tune, and amplitude of the original music. Often this creates unnatural, odd harmonics which result in the hard, fatiguing sound associated with early digital recordings. Note on the graphic below that the solid blue line represents the actual music wave and the black dots represent the closest quantization values.

Though modern sampling rates are high enough to fool the human ear, quantization errors still occur when translating from one format to another. For example, when Sony decided to archive their analog master libraries to DSD64 back in 1995, they were wrong to believe that these masters would be future-proof and able to reproduce any consumer format. The fact is, these masters could only properly reproduce a format that was divisible by 44.1KHz. So any modern 96KHz or 192KHz recording created from DSD64 master files have quantization errors.

This leads me to one of the many things that enrage me about the recorded entertainment industry. If 44.1KHz was the standard which was engineered to put aliasing errors in less critical audio frequencies, then why did they start using multiples of 48KHz?!?!?!? All they had to do was go with 88.2KHz and 176.4KHz as the modern HD consumer formats, and all of this mess could have been avoided. They made DXD, a 24-bit 352.8KHz studio format, equally divisible by 44.1KHz. What blithering idiot decided to put a wrench in the works with 96KHz and 192KHz HD audio?!?!?!?

The actual reason for the 48KHz multiple has to do with optimal synchronizing to video. So it makes sense to have sound tracks from movies recorded in a 48KHz multiple, such as the 24-bit 96KHz format embedded into 7.1 channel audio on DVDs and Blu-Rays. But since over 90% of all music recordings are sold in a 44.1KHz for Red Book CD or DSD64 SACD it is rather ridiculous to offer any HD music in 96KHz or 192KHz as opposed to the optimal 88.2KHz and 176.4KHz HD formats. But because naive consumers wrongly believe that the higher the sampling rate the higher the fidelity they demand 192Khz falsely believing it is better than 176.4KHz, so that is what record companies market.

Quantization noise is unavoidable. No matter what format you digitize in, ultrasonic artifacts are created. The more bits you have, the lower the noise floor. Noise floor is lowered by roughly 6db for each bit. So as you can imagine, 1-bit DSD has significantly more ultrasonic noise than even 16-bit PCM. This is part of why wide-DSD formats with 5-bit to 8-bit parallel Delta-Sigma decoding were created. With PCM, you have to deal with significant noise at the sampling frequency. This is why Sony and Philips engineered the Red Book CD to sample at 44.1KHz, which is over twice the human high-frequency hearing limit of 20KHz.

Since quantization noise is present around the sampling frequency of a PCM recording, a 44.1KHz recording has quantization noise one octave above the human hearing limit of 20KHz. This quantization noise needs to be filtered out, so all DACs have a low-pass filter at the output. Because the quantization noise is only one octave above audibility the filters used have a very steep slope so as to not filter out desirable high frequencies. These steeply sloped low-pass digital filters are commonly known as "brick wall" filters. This is why there can be an advantage in playing 44.1KHz PCM upsampled to 88.2KHz or 176.4KHz.

Of course DSD64 is another story: above 25KHz the quantization noise rises sharply, requiring far more sophisticated filters and/or noise-shaping algorithms. See graphic below. When you filter the output of DSD64 with a simple low-pass filter, the result is distorted phase/time and some rather nasty artifacts in the audible range. The solution is noise-shaping algorithms which move the noise to less audible frequencies and/or higher sampling rates. This is why Double-Rate DSD and Quad-Rate DSD formats came into being. This is also why advanced player software, such as JRiver, offers Double-Rate DSD output. Using player software that upsamples DSD64 to DSD128 or DSD256 significantly improves performance by putting the digital artifacts octaves above audibility allowing more advanced noise-shaping algorithms and less severe digital filters. Note these extremely high sampling frequencies are why ultra accurate clocking is more important in the playback of DSD than PCM recordings.

Jitter is defined as inconsistencies in playback frequency caused by inaccurate clocking. The result is observable as distortion of the time and tune of the music. Often the pattern of the inconsistency of frequency can result in an analog wave form that has an unnatural odd harmonic frequency. This results in the fatiguing character commonly known as “digititis.” Note in the two graphs below: jitter is an inconsistency in the horizontal time axis and non-linearity is an inconsistency in the vertical amplitude axis.

Jitter occurs when the converter’s clock rate is inconsistent and non-linearity can occur when the converter's reference voltage is inconsistent. This is why we are hearing so much about “super clocks” and “femto clocks.” The more accurate the clock, the more accurate the analog output. This is also why ultrahigh-performance R-2R DACs, such as Mojo Audio’s Mystique, have a way to adjust the voltage of the most-significant-bit (MSB) at the zero crossing to optimize linearity.

The Myth of Pure DSD:

Despite the marketing hype, there are almost no pure DSD recordings available to consumers. This is partially because up until quite recently there was no way to edit, mix, and master DSD files. So most pure DSD recordings which are commercially available are those recorded direct to DSD without any post-production. There are some new studio software packages which can edit, mix, and master in DSD, but these are quite rare in the industry, and mostly used by small boutique recording companies. Most DSD recordings are in fact, edited, mixed, and mastered in PCM and then converted back to DSD. The marketing hype DSD flow chart you see below rarely exists anywhere but in theory. Yikes…the secret is out.

There are several generations and levels of quality in purely digital DSD recordings. The least pure are DSD recordings made from old PCM masters. Many of these PCM masters had low resolution as well as significantly higher quantization errors and lower linearity than modern PCM recordings. Since you can never get better than the original masters, these DSD recordings sound as bad as or worse than the original low-resolution PCM masters. The purest common DSD recordings come from modern DSD masters which are recorded in 5-bit to 8-bit Wide-DSD, which is in fact a 5-bit to 8-bit parallel Delta-Sigma encoding.

As you can see from the above flow chart, most commercially available DSD recordings have to be converted back and forth to a PCM format in order to do post-production editing, mixing, and mastering. In each of these conversions, more quantization noise and/or quantization errors are added to the recording. For that reason they created these inaudible resolution 24-bit and Wide-DSD formats with insanely high sampling rates. The higher the resolution during editing, mixing, and mastering, the lower the digital noise in the audible spectrum when these recordings are downsampled to commercially available formats.

It is quite unlikely that any or many of recording studios that are currently using Wide-DSD for editing, mixing, and mastering will ever upgrade to software that can edit, mix, and master in true DSD, since DSD is in fact an obsolete format. Even Sony no longer supports DSD and SACD. The modern format which recording studios will likely be upgrading to would be MQA, which compresses much better than DSD or PCM for streaming and decodes to PCM formats, such as 24-bit 88.2KHz. That is why HD music streaming services such as Qobuz and Tidal are switching over to MQA for their ultra-HD selections. So with the invention of MQA compression, PCM is quickly becoming the preferred HD music format.

Another common marketing myth about DSD vs. PCM is that when blind listening tests were done comparing DSD to PCM, there was a consensus that PCM had a fatiguing quality and DSD had a more analog-like quality. This was proved to be total marketing BS. One way that marketing lie was perpetuated was with hybrid SACDs which have DSD64 and 16-bit 44.1KHz PCM on the same disk. The DSD64 tracks have over 30 times the resolution of the 16-bit 44.1KHz tracks so that they could make DSD sound better than PCM in comparisons. The truth is that in recent blind studies they've proved that high-resolution PCM and DSD are statistically indistinguishable from one another. Considering that nearly all DSD recordings were edited, mixed, and mastered in PCM, it is no wonder.

Then there are the differences in the ways DAC chips work. Most modern DAC chips are Delta-Sigma which decode native DSD. R-2R DAC chips decode native PCM. In order for you to play PCM files on a Delta-Sigma DAC or DSD files on an R-2R DAC the files have to be converted in real time.

Most modern Delta-Sigma DAC chips can decode multiple file formats, including PCM, DSD, and Wide-DSD. When they are decoding PCM, a Delta-Sigma DAC chip has to first convert it into DSD, the chip's native format. Another reason for the common misconception that DSD performs better than PCM has to do with the poor quality of the real-time PCM to DSD converters built into native DSD Delta-Sigma DACs. Since R-2R ladder DAC chips can only decode PCM formats some DAC manufacturers use chips or FPGAs at the input stages of their DACs which convert DSD to PCM. But no R-2R DAC chip can decode DSD on its own.

In almost all cases I would recommend playing music files in the native format which your DAC chip decodes. That would be PCM for an R-2R DAC chip and DSD for a Delta-Sigma DAC chip. There are several brands of player software on the market which have real-time PCM to Double-Rate DSD converters. HQ Player is one of the most sophisticated player software packages on the market today. HQ Player can be configured for real-time PCM to DSD conversion as well as real-time DSD upsampling to Double, Quad, Octuple, and even higher rate DSD formats. Using player software that is capable of converting PCM to DSD and upsampling it to at least Quad-Rate DSD is highly recommended.

Summary:

Well, all that’s a real ear opener, isn’t it?

When people claim to hear significant differences between PCM and DSD it is not the difference between the formats that they are hearing, but most often the difference in the quality of the digital remastering or the native format their specific DAC decodes. Delta-Sigma DACs decode native DSD and R-2R DACs decode native PCM.

Keep in mind that most recordings are engineered to sound best on a car stereo or portable device as opposed to on a high-end audiophile system. It’s a well-known fact that artists and producers will often listen to tracks on an MP3 player or car stereo before approving the final mix.