Yesterday I began an exhibition of the best 12AX7 tubes left in my thermionic collection.  Perhaps someday I will expand the list to include 12AU7 and 6SN7 types.  As for today, I will continue with 'Part 2' of my favorite 12AX7 still in my collection.  Also, I am going to throw in some bonus content (perhaps in 'Part 3') as I have some late 1940's General Electric/Ken-Rad 12AX7 prototypes that look suspiciously like the first to market RCA 12AX7.  Actually they look more like a hybrid of the two, and not always pretty as such!  Saving that for later, here goes with my next most favorite 12AX7 in my collection... perhaps the most underrated and unknown American made 12AX7.  The mid 1950's short black plate Sylvania 12AX7.  What a sublime sounding tube!

​In the mid-20th century, CSF (Compagnie Générale de Télégraphie Sans Fil) based in Saint-Égrève, France, and La Radiotechnique (a subsidiary of Philips) based in Suresnes, France, developed a closely knit relationship centered around the production of high-quality vacuum tubes, specifically the 12AX7S models, for military applications.  (See photo to left.. these are Philips La Radiotechnique French Military tubes made under contract for CSF.  I find it interesting they are labeled on the sales contract as both 12AX7S and 5751.) 

Both companies played significant roles in supplying the French military with essential electronic components. La Radiotechnique, identified by the military code FRS, and CSF, marked by the code FSE, collaborated extensively to meet the stringent demands of military specifications. This partnership was vital in ensuring consistency and reliability in the performance of their products.

A notable aspect of their collaboration was the identical construction of the 12AX7S tubes produced by both factories. Despite being manufactured in different locations, these tubes shared the same design and technical specifications. This uniformity was crucial for interoperability and standardization across military equipment.

The 12AX7S tubes often bore both the FRS and FSE military codes, reflecting the intertwined production processes of the two companies. Typically, the FRS code of La Radiotechnique was permanently etched into the lower part of the glass tube, while the FSE code of CSF was painted on the same tube. This dual marking underscored the cooperative efforts and mutual reliance between the two manufacturers.

The relationship between CSF and La Radiotechnique highlights a period of significant collaboration in the French electronics industry, particularly in the context of military production. By aligning their manufacturing processes and maintaining stringent quality controls, they ensured that their products met the high standards required for military use. This partnership not only facilitated the production of reliable and high-performance vacuum tubes but also exemplified the broader trend of industrial cooperation during an era of technological advancement.
​

In summary, the partnership between CSF and La Radiotechnique was a model of industrial collaboration, driven by the demands of military precision and excellence. Their shared efforts in producing identical 12AX7S tubes, marked by both FRS and FSE codes, underscore the depth and success of their cooperation.

Oh, and did I mention?  These are some of the best sounding French tubes you can ever buy.  No they are NOT 'MAZDA' tubes, although you will see some major sellers call them as such.  MAZDA tubes are only properly made by British Thomson-Houston and French Thomson-Houston and their subsidiaries such as CIFTE.  Indeed, Thomson purchased shares of CSF in the 1970's, but they simply resold the FRS code Philips La Radiotechnique tubes as explained above, therefore they are not the same as the French MAZDA tubes that most people have in mind.  

​See the photos below.  These are RT 12AX7S through and through, with FRS Suresnes factory codes.  Resold by CSF, yes, but in this case not made at the CSF factory.  Which seems to hold true for all the ones I have found.  I have yet to find the reverse; a 12AX7S of this same construction with CSF FSE codes etched permanently into the glass, with FRS La Radiotechnique simply painted on.  But, the world of French tubes is pretty wild and not for the faint of heart!  They could exist, and if anyone has one or has seen one, please, please let me know.  

​If you want to see very similar in construction, yet ACTUAL CSF of St. Egreve made tubes, stay tuned.  They are next in line! (No La Radiotechnique codes to be found anywhere on these!)

For a period of many, many years, I was an avid vacuum tube collector.  I had the 'sickness' so badly, that one day I opened my closet and counted well over three thousand tubes.  I had to do something, so I sold a great many of them, and kept the ones I just could not seem to let go of.  

Unfortunately, I let go of more than I really wished to do so.  There were some excellent tubes that went up for sale, including some of the earliest Mullard CV4004 tubes, dated into the early to mid-1950's.  I sold numerous Philips Holland Long Plate 12AX7 with the Bugle Boy graphic, all dating to the late 1950's.  And I let go of several French 'Mazda' tubes with the bright chrome plates, both of the triple and double mica type.  I even let go of some of my favorite CSF-Thomson tubes, that were apparently products of the French Military, having etched codes indicating they were made in Suresnes, France by Philips at the Radiotechnique plant, while being painted with CSF factory codes!  What a tangled web tubes can be!

All that bemoaning aside, I am beginning a multi-part series on the best sounding 12AX7 tubes (in my opinion) left in my still substantial collection.  I hope you enjoy, and maybe we all will learn a little something new. 

​

There are many ways to 'skin a cat' as we say down here in the heart of Appalachia, with the Smoky Mountains and 'Rocky Top' in direct view from my porch as I am writing this.  A truly inspiring scene to relax and then paradoxically write a mind numbing technical comparison. 

Why is DSD on my mind?  I seem to have a sickness; a truly impulsive need to explore its 'mysteries'.

Today I have chosen two 'Direct' DSD DACs to compare their approach to Digital to Analog Conversion. 

The first of these is the Signalyst DSC Discrete DSD DAC, which cannot be bought via any normal 'retail' method.  You have to build it yourself or have one built for you.  The intellectual creator is Jussi Laako of Signalyst, maker of ​HQPlayer Software. My actual build is the work of Pavel Pogodin.  You can read more by clicking here.  Before going any further, I need to point out this is a  DSD only DAC.  PCM cannot be played without conversion to DSD.  The DSC DAC is designed to be used with HQPlayer software, a powerful tool that converts PCM to DSD, and lower rate DSD to higher rates up to DSD1024.  (Note the actual DSC DAC can only accept up to DSD512.)  

The second of these DSD DACs under discussion is the PSAudio DirectStream DAC, which comes in two versions, MK1 or MK2.  The distinctions between MK1 and MK2 play no real role here; the basics are the same.  The DIRECTSTREAM will accept both PCM and DSD, and needs no external software for PCM because similar conversion functions are done internally in the FPGA brain of the DAC.  

Due to the 'separates' nature of the HQPlayer external software combined with the DSC DAC, you actually get what may, counterintuitively at first glance, be an advantage.  Although both DACs boast 'Direct' DSD processing, only the DSC DAC can actually convert DSD with no extra DSP required, because the PSAUDIO DAC has no type of DSD 'bypass' mode.  DSD via the PSAUDIO DIRECTSTEAM will always undergo DSP and is never 1-bit at all times.   

The SIGNALYST DSC DAC can actually process DSD directly with no DSP via two methods.  The first is simply connect the DAC to a PC, Network Streamer with USB output, etc., and only send it DSD signals.  The other way is to use flexible software, such as the HQPlayer software, in which the user has a choice to send DSD files directly and bitperfectly to the DAC as an additional possible path if one wishes not to use any DSP.   However, it is probably necessary to point out that the DSP systems in both HQPLAYER and in the DIRECTSTREAM (especially so in the DIRECTSTREAM) are programmed for optimal synergy with their respective hardware analog conversion systems.  

I do not want to delve too much more into the DSP side of things, because I want to focus here on the actual conversion of DSD itself, after any DSP.  But the reason the DSP exists is to provide extremely well designed oversampling filters for both PCM and DSD, which can be more capable and accurate than what you find in a typical DAC.  It also allows for digital volume control, even on DSD.  

Once things get past the DSP, things really get a bit more similar.  Both use discrete analog components to filter their DSD signal (Reminder, no matter what signal you put in either system, either PCM or DSD, it will be internally converted to DSD for analog conversion.)  The DIRECTSTREAM claims to be a completely passive system; however the SIGNALYST DSC uses a Sallen-Key Filter as part of its DSD filtering, which is an active component.  

So, how do they work??  Here comes the fun part.  Or the part where many of you may tune out.  Technical stuff is ahead.  You have had your fair warning :) 

Let's start with the Signalyst. 

It uses what I would call a more traditional and common method of DSD conversion.  Most DACs, even though they are not made of as many discrete components, use something similar IF they have a bypass or native DSD mode.  

Disclaimer.  These descriptions are a bit generalized.  They may not take into account every possible step or piece of hardware.  

The SIGNALYST DSC starts by receiving the 1 bit DSD signal, which can be as high as DSD512, and sends it into a discretely built analog filter.  The filter uses typical digital techniques, but is implemented in the analog domain.  Some may see this as a Digital/Analog Hybrid filter.  The filter is a type of CIC filter, with FIR filter characteristics, that excludes the decimation stage.  It just filters and smooths samples into analog.  It doesn't discard any actual samples in the process.  

The way this works is pretty darn ingenious.  What you need are shift registers (flip-flops, no not the sandal), a MOSFET or transistor to act as a switch that either connects or disconnects an 'output element' resistor (which is the equivalent of a digital filter TAP), and some kind of summation node. 

CIC FILTER

1. Shift Register:
   • Function: The shift register takes the incoming DSD bitstream and creates multiple parallel bitstreams, each offset by one clock cycle. In the case of the SIGNALYST DSC DAC, it takes 32 consecutive bits of the DSD stream, and places the 32 identical DSD bitstreams stacked upon one another, and remember each stream is offset by one tick of the bit-clock.  Due to the nature of this kind of bit coding (thermometer code), it actually means 33 conversion levels, because we can't forget about 0 level.  The timing offset out of the shift register creates a smoothing, comb filter effect when combined with the output elements (voltage controlled resistors) and final summation of all signals into one again.      
2. Voltage-Controlled Resistors and Switches:
   • Role: Each bitstream controls a switch (e.g., a MOSFET or a transistor) that either connects or disconnects a resistor from the summation node.
   • Design: The resistors can be chosen to provide a weighted contribution to the summation based on the timing of the bitstream. In the case of the DSC DAC, however, the element are of equal weighting, so there is no further contribution to the filtering.  
3. Analog Summation:
   • Summation Network: The analog outputs of the switches are summed together. This summation integrates the 1-bit DSD bitstream into a continuous analog signal.
   • Filtering: The combined effect of the time-offset bitstreams, output elements, and the summing network completes the low-pass filter, smoothing the high-frequency components and leaving a clean analog signal.

But this still isn't quite enough filtering for the extremely high levels of ultrasonic noise.  It DID accomplish conversion from digital to analog, and did a great deal of filtering itself, but we need more.  

The SIGNALYST DSC DAC follows the CIC Comb filter that converted the digital signal to analog with another analog filter.  One that assists in further shaping out that ultrasonic noise.  In comes some active filtering.. a Sallen-Key filter. 

SALLEN-KEY FILTER

• Additional Filtration: The Sallen-Key filter will provide further low-pass filtering to ensure that any high-frequency artifacts not completely attenuated by the passive filter are removed.
• Signal Conditioning: It can also help in signal conditioning, providing a clean and smooth analog output.  (It is worth noting that some other similar designs stay passive with an RC filter in this position instead. (iFI Audio, I am talking about you-- also note iFi while using a very similar conversion technique, uses unequally weighted elements and has a bitstream that is only 8 bits long.)  

By following the passive discrete component hybrid digital/analog CIC output filter with an active Sallen-Key filter, we achieve a robust and comprehensive filtering solution for converting DSD to a high-quality analog signal. This combination leverages the strengths of both passive and active filtering techniques, ensuring minimal high-frequency noise and excellent signal integrity.

FINAL STAGE: OUTPUT TRANSFORMER

While most other DACS I can think of use different kinds of final analog output methods, both the SIGNALYST DSC DAC and the DIRECTSTREAM DAC have chosen to use output transformers.  

Purpose:

• Impedance Matching: Ensures the output impedance matches the input impedance of the next stage (e.g., an amplifier or audio interface).
• Isolation: Provides galvanic isolation to reduce ground loops and noise
• Signal Smoothing: Further smooths the signal by filtering out any remaining high-frequency components.

By following this design approach, the SIGNALYST DSC DSD DAC achieves high-fidelity conversion of DSD to analog, leveraging the strengths of discrete components and innovative filtering techniques.  One thing to note before we move on to the DIRECTSTREAM, is the CIC filter by its natural design will change its filter cutoff frequency with each change of DSD speed, and it will double with the change.  DSD64 may hypothetically start its rolloff at 30khz.  DSD128 would start at 60khz, DSD256 would start at 120khz, etc.  This is important for later comparison with the DIRECTSTREAM DAC.

Moving on to the PSAudio DIRECTSTREAM DAC

This one is unique.  I know of no other DAC currently available that uses this technique.  The previously discussed technique used in the SIGNALYST DSC DAC is quite standard across the industry, and the schematics for it are Open Source, so its easy to get to the details of operation. Not so here.  We have some major differences, derived from a few clues thrown our way.  Because it's proprietary intellectual property, expect a shorter and less deep dive into its operation. 

The DIRECTSTREAM, as mentioned in the beginning, contains its own bespoke digital filters and digital volume control on its FPGA.  The 'intermediate signal' where the Volume Control, Balance Control, and whatever other DSP it uses, is at least at 30bit per sample signal at least 10x the DSD64 rate.  This minimum 30bit ultra-high sample rate signal is used for DSP on both PCM and DSD.  

They advertise this is always a 1-bit system that never is converted to PCM. ​ I find this misleading and inaccurate.  What they are trying to say is, when a 1-bit PURE DSD SIGNAL is input into the DAC, the signal isn't ever decimated to any type of low PCM rate.  The truth is, both PCM and DSD use an interpolation filter.  Once DSD is interpolated, it is no longer a 1-bit, time splicing noise shaped signal, although of course this 'intermediate' signal can be oversampled and re-noise shaped into whatever bit depth and sample rate one could want.  

The actual DSD signal is oversampled by probably an FIR filter, just like Sony DSD-Wide of old, and ESS Sabre of today, into a huge 30 bit 28.224 MHZ signal!! (MKI) 

(Nothing new is under the sun, and there is no 'magic' in how DSP is applied to 1-bit DSD.  Even now, decades later, the best DSD recording systems are using the same techniques as yesteryear.  Many  are seeming to stay with a Sony 'DSD-Wide' type approach.)

Why a signal so big?  Well, one reason would be you can use tremendously large digital FIR filters with extreme accuracy and control, by being able to implement millions upon millions of filter TAPS.  This is evidenced by the impulse response measurements that have appeared in the big pro magazines when the PSAUDIO DIRECTSTREAM DAC is on their test bench.  The filter rings seemingly forever, reminding me of a Chord product.  Yes, there are advantages here, but, all that ringing is a major disadvantage.  But that is a subject for a different day.  Additionally, with DSD material, the FIR oversampling filter could help with ultrasonic noise control as it will pre-filter the signal in this multi-bit stage.  

After the DSP is finished, it uses a delta-sigma modulator to convert everything to DSD128.  Remember how the SIGNALYST DSC outputs multiple rates that change the filter characteristics?  That doesn't happen here.  Everything is converted, PCM and  DSD no matter what the rate, even DSD 256, to DSD128.  WHY?  That is something more than this article can cover, but there is the idea of a DSD 'sweet spot' where extra speed is actually detrimental to the sound and makes for a more difficult analog conversion, counterintuitively at first, until you understand why.  I suggest you read Andreas Koch talk about it here.  

So we make it to our final bitstream. 1 bit DSD128.  This is where the fun begins (again)....

Instead of the more common discrete CIC (FIR) filters used to convert 1 bit DSD to analog, the DIRECTSTREAM uses something I would never have thought of... a class A video amplifier!!!

Using a Class A video amplifier to filter a DSD bitstream is a quite sophisticated method to achieve high-fidelity audio output. This approach leverages the high-speed and wide bandwidth capabilities of video amplifiers, which can handle the high-frequency components of DSD signals effectively.

CHARACTERISTICS

1. High Bandwidth:
   • Video amplifiers typically have very high bandwidth, well into the MHz range, which is essential for handling the high-frequency content in DSD signals (e.g., DSD64 at 2.8224 MHz).
2. Low Distortion:
   • Class A operation ensures low distortion and high linearity, which is critical for maintaining the integrity of the audio signal.
3. High Slew Rate:
   • The ability to respond quickly to changes in the signal makes video amplifiers suitable for the fast transitions present in DSD bitstreams.

MORE CONCEPTUALIZATION

1. Direct Amplification:
   • The DSD bitstream is directly fed into the Class A video amplifier. The amplifier’s particular bandwidth allows it to pass the high-frequency components that are to be kept, and to attenuate those that need to be discarded.  
2. Low-Pass Filtering:
   • By leveraging these frequency response characteristics of the amplifier and possibly additional passive components, high-frequency noise can be filtered out, leaving a clean analog signal.

CIRCUIT DESIGN

1. Input Stage:
   • The input stage receives the DSD bitstream and prepares it for amplification. This stage needs to be designed to match the impedance of the bitstream source and ensure proper signal levels for the amplifier.
2. Amplifier Stage:
   • A high-bandwidth Class A video amplifier must be used. These amplifiers provide the necessary speed and linearity.  The output of the amplifier stage is fully analog.  The digital bitstream has now been converted into a filtered higher voltage analog representation.  
3. Output Filtering:
   • An RC (resistor-capacitor) network can be used at the output to provide additional low-pass filtering. This network can help to further smooth the signal by attenuating frequencies above the audible range.

ADVANTAGES

1. High Fidelity:
   • The use of Class A video amplifiers ensures high fidelity due to low distortion and high linearity.
2. Wide Bandwidth:
   • Capable of handling the high frequencies associated with DSD bitstreams.
3. Simplicity:
   • Simplifies the filtering process by leveraging the inherent characteristics of the video amplifier.

We are not finished yet.  Just like the SIGNALYST DSC DAC, the DIRECTSTREAM uses an output transformer for the same exact functions.  It offer some filtration to go along with the filtration of the Video Amplifier, along with possible other passive analog filtering such as an RC filter.  Often this  output transformer filter function is referred to as working at DSD256. 

That made no sense to me at first.  I first saw it stated that way in Hi-Fi News.  But, the transformer doesn't put out "DSD256" by any means, nor any other bitstream.  It is an analog signal at this point.   

What is happening here is this: the previously discussed full analog system (sans transformer) is designed for conversion and filtering of one rate: DSD128.  But the TRANSFORMER is optimized for DSD256 filtering.  Here is why:

​If the output transformer, which has its own low pass filter capabilities, is optimized for DSD256 and is used with a DSD128 signal, it will have a higher cutoff frequency. This approach helps to preserve the transient response of the signal. Here’s a detailed explanation of why this is the case and the implications for audio performance:

Output Filter Optimization

1. Filter Cutoff Frequency:
   • DSD128 vs. DSD256: DSD128 has a sampling rate of 5.6448 MHz, while DSD256 has a sampling rate of 11.2896 MHz. A filter optimized for DSD256 would typically have a cutoff frequency that is suitable for handling the higher frequency noise components associated with the higher sampling rate.
   • Higher Cutoff Frequency: When this filter is applied to a DSD128 signal, the higher cutoff frequency allows more high-frequency content to pass through, which can improve the transient response of the audio signal.
2. Transient Response:
   • Preservation of High-Frequency Details: A higher cutoff frequency means that more high-frequency transients and details are preserved in the analog output. This is crucial for maintaining the clarity and accuracy of fast, transient-rich audio signals.

Practical Implications

1. Noise Shaping and Filtering:
   • High-Frequency Noise: DSD signals inherently contain high-frequency quantization noise. Filters optimized for higher rates (like DSD256) are designed to attenuate this noise effectively without impacting the audible range.
   • Application to DSD128: When such a filter is applied to a DSD128 signal, it will still attenuate high-frequency noise but may do so less aggressively than a filter specifically designed for DSD128. This results in a cleaner transient response but may allow some high-frequency noise to remain.
2. Audio Quality:
   • Enhanced Detail and Clarity: By preserving more high-frequency content, the audio output can benefit from enhanced detail and clarity, particularly in complex and dynamic recordings.
   • Potential Trade-offs: There is a balance to be struck between preserving transient response and minimizing high-frequency noise. The design of the filter must consider this to optimize overall audio performance.
   •  The filter designed for DSD256 will attenuate less aggressively, preserving more high-frequency content and improving transient response, at the cost of potentially letting through more high-frequency noise.

Optimizing an output filter for DSD256 and using it for a DSD128 signal results in a higher cutoff frequency, which helps in preserving transient response. This approach enhances the clarity and detail of the audio signal but must be carefully balanced to manage high-frequency noise effectively. This method highlights the importance of considering the specific characteristics of both the signal and the filter in high-fidelity audio design.



So now let's look at a direct comparison between the SIGNALYST DSC DAC and the PSAUDIO DIRECTSTREAM DAC, highlighting relative strengths and weaknesses. 

Method 1: Class A Video Amplifiers for Filtering DSD Bitstream
Characteristics:

• High Bandwidth: Class A video amplifiers can handle the high-frequency content of DSD signals effectively.
• Low Distortion: Class A operation ensures low distortion and high linearity.
• High Slew Rate: Suitable for fast transitions in DSD bitstreams.

Implementation:

• Direct Amplification: The DSD bitstream is directly fed into the Class A video amplifier.
• Output Filtering: Possibly uses RC networks for additional low-pass filtering to remove high-frequency noise.
• Output Transformer: Ensures impedance matching, signal isolation, and additional smoothing.

Advantages:

• Simplicity: Direct approach with fewer components.
• High Fidelity: Low distortion and high linearity provide excellent audio quality.
• Wide Bandwidth: Capable of handling high-frequency components inherent in DSD.

Disadvantages:

• Power Consumption: Class A amplifiers dissipate a lot of power and require good thermal management.
• Limited Filtering: While video amplifiers have wide bandwidth, they may not provide as precise filtering as dedicated filter circuits.

Method 2: Discrete Component CIC Filter Plus Sallen-Key Filter and Output Transformer

Characteristics:

• CIC Filter: Uses shift registers and resistor networks to create a composite analog signal from the DSD bitstream.
• Sallen-Key Filter: Provides precise low-pass filtering to remove high-frequency noise.
• Output Transformer: Ensures impedance matching, signal isolation, and additional smoothing.

Implementation:

• CIC Filter: Shift registers create multiple parallel bitstreams, each processed through resistors and then summed.
• Sallen-Key Filter: Active low-pass filter with high precision.
• Output Transformer: Final stage for signal conditioning and isolation.

Advantages:

• Precision: Sallen-Key filters provide precise control over the filtering characteristics.
• Comprehensive Filtering: Combined stages ensure thorough removal of high-frequency noise.
• Signal Conditioning: Output transformer adds benefits of impedance matching and isolation.

Disadvantages:

• Complexity: More components and stages involved.
• Size and Cost: Potentially larger and more expensive due to the number of components. (from a raw parts perspective- not a retail perspective)
• Active Components: Requires power and careful design of active filter stages.

COMPARATIVE ANALYSIS

1. Filtering Precision:
   • Class A Video Amplifiers: Good for general filtering with high fidelity but may not achieve the same level of precision in filtering high-frequency noise as the discrete component approach.
   • Discrete Component Approach: Offers more precise and controlled filtering, particularly effective in removing high-frequency noise due to the combination of CIC and Sallen-Key filters.
2. Complexity and Power Consumption:
   • Class A Video Amplifiers: Simpler with fewer components but higher power consumption and heat dissipation requirements.
   • Discrete Component Approach: More complex and larger, with additional power requirements for active components, but generally more efficient in specific filtering tasks.
3. Audio Fidelity:
   • Both methods can achieve high audio fidelity, but the discrete component approach with Sallen-Key filtering might offer better overall noise reduction, especially for high-end audio applications where precise filtering is critical.
4. Implementation and Cost:
   • Class A Video Amplifiers: Easier to implement with fewer components, potentially lower cost for simpler designs.  (not from a retail perspective)
   • Discrete Component Approach: Higher complexity and cost but provides a more comprehensive filtering solution. (not from a retail perspective)

CONCLUSION
The choice between using Class A video amplifiers or a discrete component CIC filter plus Sallen-Key filter with an output transformer depends on the specific requirements of the application:

• For simplicity and high fidelity with high bandwidth capability: Class A video amplifiers are a good choice, especially if power consumption and thermal management can be handled.
• For precise and comprehensive filtering: The discrete component approach with CIC and Sallen-Key filters, followed by an output transformer, offers superior noise reduction and signal conditioning, making it ideal for high-end audio applications.

  

The perception that audio systems sound better at night is a common experience among audio enthusiasts. I am among those.

While you will never, never, ever have an easy time convincing me that a 5,000 US dollar RCA interconnect is making ANY noteworthy difference, I think there is a real case to be made for this nighttime audiophile 'tale'.  I am a believer.  

While it may seem subjective, there are several technical reasons why this could be the case. Here are some possible explanations:
​

1. Reduced Electrical Noise

• Lower Power Demand: At night, the overall demand on the electrical grid is lower because fewer electrical devices are in use. This can lead to cleaner power with fewer fluctuations and less electrical noise.
• Improved Power Quality: The reduction in industrial and commercial electrical activity at night can result in fewer disturbances and less interference on the power lines, providing a more stable and noise-free power supply to your audio equipment.

2. Reduced Radio Frequency Interference (RFI)

• Less RFI at Night: Many sources of radio frequency interference, such as industrial equipment and office electronics, are typically turned off at night. This reduction in RFI can lead to a quieter background, allowing audio systems to perform better.
• Cleaner Signal Path: With fewer electronic devices operating, there is less chance for RFI to be introduced into the audio signal path, resulting in clearer sound reproduction.

3. Ambient Noise Levels

• Quieter Environment: At night, the ambient noise levels are generally lower. There is less traffic, fewer people moving around, and overall reduced background noise. This quieter environment can make subtle details in the music more noticeable and enjoyable.
• Psychological Effect: The reduced ambient noise can also have a psychological effect, making listeners more relaxed and attentive, which can enhance the perception of sound quality.

4. Temperature and Humidity

• Environmental Conditions: The temperature and humidity levels can be different at night, which can affect the acoustics of the room. Cooler temperatures and stable humidity levels can improve sound propagation and absorption characteristics in the listening environment.
• Equipment Performance: Audio equipment, especially analog components like tube amplifiers, might perform slightly differently under varying temperature conditions, potentially affecting sound quality.

5. Reduced Electrical Interference from Other Appliances

• Fewer Appliances in Use: Many household appliances that create electrical noise (like refrigerators, air conditioners, and washing machines) are typically off or used less frequently at night. This reduction in appliance-related electrical noise can improve the performance of audio systems.
• Dedicated Power Supply: With fewer devices drawing power, the audio system may benefit from a more dedicated and stable power supply, reducing the risk of voltage drops and power line noise.

While the perception of better sound quality at night can be influenced by subjective factors, there are several technical reasons that support this phenomenon:

• Reduced Electrical Noise: Lower power demand and cleaner power at night.
• Reduced RFI: Fewer sources of radio frequency interference.
• Quieter Environment: Lower ambient noise levels.
• Optimal Environmental Conditions: Potentially improved acoustics and equipment performance due to temperature and humidity.
• Fewer Competing Appliances: Less electrical interference from household appliances.

​These factors combine to create a more favorable listening environment, allowing audio systems to perform at their best and enhancing the overall listening experience at night.

Pseudo multi-bit Delta-Sigma Modulation (DSM) is a technique used in digital audio processing to improve the performance of digital-to-analog and analog-to-digital converters. It involves using multiple 1-bit quantizers to mimic the behavior of a single multi-bit quantizer, combining their outputs to achieve better signal quality and noise shaping.


Key Concepts

1. Delta-Sigma Modulation (DSM):
   • A method used to convert analog signals to digital (and vice versa) by oversampling and shaping quantization noise to higher frequencies where it can be easily filtered out.
2. Pseudo Multi-Bit DSM:
   • Utilizes multiple 1-bit quantizers to approximate the performance of a single multi-bit quantizer.
   • Combines the outputs of these 1-bit quantizers to create an effective multi-bit output.
3. Selective Feedback:
   • Involves selectively feeding back certain 1-bit quantizer outputs or combinations to optimize noise shaping and linearity.
4. Unary (Thermometer) Code:
   • A unary code represents each quantization level with a specific number of 'on' elements in a sequence, ensuring monotonicity and improving linearity.
   • Conversion to a thermometer code helps in effectively managing and processing the quantizer outputs.
5. Dynamic Element Matching (DEM):
   • A technique used to average out mismatches in the digital-to-analog conversion process by dynamically reordering the active elements.
   • Helps in reducing distortion and improving the overall linearity and performance of the modulator.

How It Works
​

1. Input Signal Processing:
   • The analog input signal is oversampled, increasing its resolution.
   • Noise shaping filters process the oversampled signal to spread quantization noise across a wider frequency range.
2. Multiple 1-Bit Quantization:
   • The processed signal is fed into several 1-bit quantizers operating in parallel.
   • Each quantizer produces a single-bit output based on the input signal.
3. Combining Quantizer Outputs:
   • The 1-bit outputs are combined using digital logic to form a pseudo multi-bit output.
   • This can involve a binary weighted sum or conversion to a thermometer code (unary representation) for further processing.
4. Selective Feedback Path:
   • Instead of feeding back the combined output directly, specific 1-bit outputs or their combinations are selectively fed back into the noise shaping loop.
   • This selective feedback can be dynamically controlled to optimize performance under varying signal conditions.
5. Dynamic Element Matching (DEM):
   • The combined outputs, if not already in unary/thermometer code, are converted to such by digital logic in the DEM circuitry.
   • DEM dynamically reorders the active elements to average out mismatches and reduce distortion, ensuring better linearity.  

Benefits of Pseudo Multi-Bit DSM

1. Improved Signal-to-Noise Ratio (SNR):
   • By using multiple quantizers, the system reduces quantization noise, resulting in a cleaner signal.
2. Enhanced Linearity:
   • Selective feedback and dynamic element matching (DEM) techniques help maintain linearity and reduce distortion.
3. Adaptive Performance:
   • The ability to dynamically adjust the feedback path allows the system to adapt to different signal conditions, maintaining high performance across various scenarios.
4. Cost-Effective:
   • Provides many benefits of true multi-bit systems without the higher complexity and cost associated with precise multi-bit DACs and ADCs.

Applications

Pseudo multi-bit DSM is widely used in high-fidelity audio systems, professional audio equipment, and precision measurement instruments where maintaining high signal quality is crucial.


Conclusion

Pseudo multi-bit DSM is a sophisticated technique that enhances digital audio processing by using multiple 1-bit quantizers to achieve the benefits of multi-bit systems. Through various possible options such as selective feedback, dynamic element matching (DEM), and the use of unary (thermometer) codes, the choice and uses of which depend on if the DSM is pure digital, or involves conversion to and from analog, it provides improved signal quality, reduced noise, and better linearity. This makes it ideal for high-performance audio applications. By understanding these principles, audio enthusiasts can appreciate the advanced technology behind modern digital audio converters.

However as effective as such techniques may be in linearizing Delta Sigma Conversion, it is 'pseudo' for a reason.  For even better performance, there is actual multi-bit Delta Sigma, which can somewhat blur the lines between Delta Sigma and PCM, but Multi-Bit DSM retains certain distinctions, especially if it makes use of unary/thermometer coding as opposed to binary coding.  The most advanced of Multi-Bit Delta Sigma converters these days, that use 64, 128 or possibly even more levels, simply must use binary (often two's complement)  for the sake of practicality.  This DOES introduce PCM 'weaknesses' if you will, into the Delta Sigma system, however, the system as a whole as judged by the incredible resolution and linearity of the latest chips from ESS and AKM, speaks for itself.  However, it is worth noting that, ironically, for the final stage of conversion, even these systems convert back into an unary coded system and use dynamic element matching to overcome the inherent linearity issues of PCM.  (See ESS 'revolver' technology and DCS 'Ring DAC' technology)

Here is an image of one of my more 'interesting' tubes.  It is a Ken-Rad GE, very clearly marked as a 5751, but with a date code from late 1950!  I have another one with same construction, but the date code isn't as clear.. it could either be from 1950 or 1951. 

1. ​Date Code: The visible date code says "0-48", which can be interpreted as follows:
   • "0": The first digit typically indicates the year of manufacture. In this case, "0" likely stands for 1950.
   • "48": The second part usually indicates the week of the year. So "48" would mean the 48th week of 1950.  
2. Tube Type: The tube type marking "5751" is clearly visible.

Given that the tube is marked as a "5751" and the date code suggests it was manufactured in 1950, this provides a strong indication that this tube might indeed be an early production model or a prototype of the 5751. Here are some key points to consider:

1. Early Production or Prototype:
   • It's plausible that this tube is part of an early production run or even a prototype batch made before the official market introduction around 1951. Manufacturers often conducted limited production runs for testing and evaluation purposes.
2. Historical Significance:
   • If this tube is indeed from 1950, it could be quite rare and significant, representing one of the earliest examples of the 5751 tube. This would make it a valuable item for collectors and enthusiasts.

This Ken-Rad 5751 tube dated from 1950 is likely an early production model or prototype, making it a rare and significant piece. This aligns with the historical introduction of the 5751 around 1951, suggesting that early development and testing were underway in 1950.

​


More photos of the tubes in question... ( again,  I have two examples)

I saw a website or  a blog that contended that ringing artifacts mean nothing because they occur at the Nyquist limit which we cannot hear, then proceeded to prove his/her point by filtering out the Nyquist frequency.  Hmmm.. sounds a lot like what Meridian and others were already experimenting with long long ago.  

However...

The perception of ringing artifacts in digital audio is more complex than simply what is audible/visible in the frequency domain. Our ears are highly sensitive to transient effects in the time domain, and even if the ringing is not prominent in the frequency spectrum, it can still impact the perceived quality of audio. Here's an exploration of this concept:

Sensitivity to Transient Effects

1. Human Auditory System:
   • Temporal Resolution: The human ear has excellent temporal resolution and can detect very small timing discrepancies in sound. This makes us sensitive to transient effects, such as pre-ringing and post-ringing, which can smear or blur the clarity of audio signals.
   • Perception of Impulses: Our auditory system is particularly adept at detecting changes and impulses in sound. Ringing artifacts, especially pre-ringing, can disrupt the naturalness of transient sounds, such as percussive hits or sharp onsets.
2. Time-Domain Artifacts:
   • Pre-Ringing: Pre-ringing occurs before the actual sound event, which is unnatural because in the physical world, effects typically follow causes. Pre-ringing can make transients sound less crisp and more smeared.
   • Post-Ringing: Post-ringing follows the actual sound event and can cause echoes or prolonged decays that are not present in the original signal. While less objectionable than pre-ringing, it can still affect the perceived clarity of the audio.

Frequency Domain vs. Time Domain

1. Limitations of Frequency Analysis:
   • Spectral Masking: Ringing artifacts might be masked in the frequency domain by other spectral components, making them difficult to detect visually in a frequency spectrum.
   • Transient Analysis: The frequency domain analysis does not always reveal transient details, as it provides an average view over a period, potentially overlooking short-term events.
2. Importance of Time-Domain Analysis:
   • Waveform Inspection: Inspecting the waveform in the time domain allows for the identification of ringing artifacts that occur around transients. This is crucial for understanding the impact on the perceived sound.
   • Impulse Response: Examining the impulse response of a filter or audio system can reveal pre-ringing and post-ringing directly. Impulse response analysis provides insights into how the system handles sudden changes in the signal.

​Mitigating Ringing Artifacts

1. Filter Design:
   • Minimum-Phase Filters: These filters shift energy to minimize pre-ringing by concentrating the impulse response after the transient event. This makes the artifacts less perceptible to the human ear.
   • Apodizing Filters: Designed to smooth transitions and reduce spectral leakage, apodizing filters can help in minimizing both pre-ringing and post-ringing effects.
2. Windowing Techniques:
   • Appropriate Windowing: Using suitable windowing functions during digital signal processing can reduce artifacts. Windowing shapes the signal to minimize abrupt changes that cause ringing.
   • Adaptive Filtering: Adaptive filters that adjust based on the signal content can better preserve transients while reducing artifacts.
3. Quality Assurance:
   • Listening Tests: Ultimately, listening tests are crucial for evaluating the perceptual impact of ringing artifacts. Human listeners can provide feedback on the naturalness and clarity of transients, guiding the refinement of filtering techniques.
   • Hybrid Analysis: Combining both time-domain and frequency-domain analyses ensures a comprehensive understanding of the artifacts and their impact.

Conclusion

While frequency domain analysis is valuable, it is not sufficient on its own to ensure high audio quality. The time domain analysis is critical for identifying and mitigating ringing artifacts that affect transients. Human ears are highly sensitive to these artifacts, and designing filters that minimize them is essential for maintaining the naturalness and clarity of audio. By addressing both domains, we can achieve a more accurate and pleasing sound reproduction.