In this entry I will be posting more measurements that didn't make it over to the main review.  I left out the filter information for PCM, because I have already measured many similar AKM DACs with the same filter profiles, so it is redundant data, if you have been checking out any of those other reviews.  (Topping E70V, SMSL D400 PRO).  

I hope you find it useful!  I have not taken a lot of time to edit here, so its a bit raw, kind of like what you get in a Blu-Ray extra feature!

Filter 5 not shown.  It is a non-oversampling filter that has little to no ringing.  My measurement ADC produces more ringing than the actual filter, therefore I am choosing to not include NOS filter impulse response graphs.  

I find lots of things in other blogs that are in varying degrees of error: normally I pass it by.  Sometimes I just cannot help it.  I must admit that there are many, many people who have forgotten more about audio than I have ever known; all the same, sometimes I might just actually know a few things about the particular subject and would like to make myself useful.   There are people who have very kindly done the same for me, offering genuine, heartfelt constructive criticism.  I would like to return that favor, and do it in similar fashion.  

Then there are others who, well, are what I call assumers.  And you know what they say about people who 'ass'ume things.  And I have had at times these 'ass'umers try and correct me, and I am astonished by how wrong they are in addition to how little they actually seem to know about the subject in question.  Genuine, heartfelt constructive criticism?  No, I do not have any of THAT for them.  

In this blog we will get a taste of both sides of me.  A couple of critiques that are in good faith, and one that, well, might me a bit more spicy.  

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THE MYSTERY OF THE VANISHING AUDIO


The first critique will be short and sweet, and honestly, I do not have a memory of what the poor fellow's name is, nor the site on which he posted this.  We are talking about Vinyl playback versus digital, and why Vinyl sounds 'better' in his mind. 

Disclosure:  I LOVE good vinyl playback myself.  I do at times think it sounds 'better' in many ways than digital.  It also obvious has some weaknesses that are pretty much inarguably large compared to digital.  And that brings me to what I consider one of the more fundamental issues in our hobby.  We always talk about why something is 'better', or 'sounds better', or just flat out 'IS BETTER.'   NO, NO, NO.  I think what we are talking about is things sound DIFFERENT.  And there are times when different people in different situations will have different PREFERENCES for what DIFFERENT they prefer.  It is a purely subjective thing in many cases.  And that is the case, I personally believe with Vinyl. 
 
In this case though, said person in goodwill stated that Vinyl (and analog in general) is better because it reproduces 'all of the audio in continuous fashion, while digital sampling leaves part of the music missing due to sampling gaps'.   FACEPALM.  Go ahead, do it with me.  Let's facepalm together.  Nothing could be more wrong.  

Digital audio in no ways 'leaves part of the music missing'.  The sampling theorem will accurately reproduce the ENTIRE waveform, within certain boundaries of frequency and amplitude accuracy.  Considering that digital has a SNR much higher than vinyl, we can go ahead and throw out amplitude accuracy as any kind of advantage vinyl may have.  So we turn to sampling rate.  

Yes, it is true that the sampling rate will limit the high frequency extension of a digital recording.  It is also true that vinyl has high frequency extension limits as well, and not only that, much, much higher distortion at the highest of frequencies.  But back to the idea of what I am going to characterize as 'holes' in the music.  Don't you also feel that is what this person is getting at?  Sampling leaves 'holes' or 'gaps' in the waveform?  Again, this cannot be any farther from the truth.  Because of the reconstruction filter.  For when the system is bandwidth limited, a proper filter will allow only ONE way for that waveform to be reconstructed from the samples.  EXACTLY as it was before it was sampled below what we call the 'Nyquist' limit.  Again, the ONLY errors that should exist below that Nyquist limit are the amplitude quantization errors, and we have already established at 16 bit and higher, they are already smaller than any amplitude distortions present in vinyl playback. 

So no, good sir, there are no gaps in digital audio.  This is a persistent myth that just will not go away for some reason. 

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THE MASTER SWITCH NOT BEING SO MASTERFUL 

I like 'The Master Switch' audio website.   I enjoy their reviews.  Pretty darn good stuff.  But I was reading their explanations of audio formats, then I got to DSD.  They were doing an okay job, until I got to this part.  I hope it's okay to use this small excerpt as fair use:

"Imagine a ruler with 44,100 lines on it. In other words, you can measure something in 44,100 increments. If the bit depth is sixteen, you’ll then be able to gather sixteen bits of information from the segment you’ve just measured. But if you have a ruler with 2,822,400 lines on it, then obviously you’ll be able to take much finer measurements. When you’re taking measurements that fine and that accurate, you simply don’t need sixteen bits of information. You only need one.
That’s because the segment you’ve measured won’t be all that different from the ones to the left and right of it. Having sixteen bits of information won’t be any more beneficial than one bit, in this case. When the sample rate is that high, there’s no benefit to having a higher bit depth."

CLICK HERE FOR TO READ THE REST AT THE MASTER SWITCH

Although the first part is extremely basic and sort of on the right track, their explanation essentially sounds like the way Delta modulation works. We are totally missing the Sigma it seems.  And the last couple sentences stating 16 bits of info is no better than 1 bit at these kind of sample rates, made me sit up and wonder it they have ever heard of multi-bit delta sigma?  (Well first, they need a primer on what Delta modulation is, but I will leave that for some one else.)  

Virtually every DAC chip in current use has a multi-bit Delta-Sigma modulator (and reminder, DSD is nothing more than 1 bit Delta-Sigma modulation stored in a bitstream file format), so OBVIOUSLY there is a major benefit to having higher than 1 bit sample rates at over 2.8 MHZ.  Actually, the latest, greatest chips are using more like 6 to 8 binary bits at rates that at times exceed 10 MHZ!  It is a way to minimize the pulse quantization error from the beginning, meaning much, much easier noise shaping requirements, and much less strain on analog output stages, not to mention massively higher resolution, both actual (from the basic principle of pulse averaging) and perceived (from the magic of noise shaping).  

Furthermore, it's not nearly as MASSIVE an increase in pulse resolution as they make it out to be.  If you take that 44.1khz sample period they are talking about, and truncate it from 16 bits to one, and consider a single sample out of that period, that is going from 65,536 levels of data in that single sample period of around 22 microseconds, to 64 individual one bit pulses/ 65 levels of data in 22 microseconds, or 6 bits when averaged.  (yes, yes, I know DSD doesn't use time periods like this to calculate its resolution, and the actual resolution changes with the frequency being sampled vs. the time period chosen in this thought experiment, but it IS a time splicing AVERAGING pulse format, and BEFORE noise shaping comes in to save the day increasing the apparent resolution by not getting rid of the error, but rather shifting it into clumps of noise at high frequencies we cannot hear, well tough.. this is accurate as to how it works.)

Expanding our horizons beyond our limited view down to an approximately 22 microsecond 44,100khz single sample, we will find a much, much greater increase in actual and perceived resolution across the entire audible range.  

FINALLY LETS GET JUICY ABOUT DITHER....

I made a simple post on the science section of a popular headphone enthusiast site the other day.  We won't talk about the real scandal 'there' that has me steamed, and that is how they treated a major vendor and massive contributor, but between their actions involving him, and the own attacks I have received there myself, (I was actually threatened by a stalker there a few years ago via PM, who hurled all manner of insults about my lack of intelligence, then proceeded to threaten to 'get' me at work, after which I actually dealt with massive amounts of A/V sabotage, in addition to stolen equipment from our normally secure audio/visual booth) and the other day a random guy in the audio 'science' section ( not a new stalker as far as I know so no worries lol) who seemed to assume I was a village idiot was just the cherry on top.. for THIS week that is.  Who knows what else will go down over there.  

This dude actually tried to tell me that DSD noise isn't quantization noise.  Rather that it is dither noise.  (As an aside, never use Gemini AI for any accurate info.  When I entered the query about the nature of DSD noise and dither, Gemini gave me his statement verbatim.  Then I looked at what source Gemini has used to come up with this info.  I can't make this crap up.  The source?  Was the very thread from the Science section of this website where this guy made the statement.  I am still laughing about the ridiculousness of this.)

Anyway, NO 1-bit DSD is NOT dither noise.  Yes, it is noise, but it is almost entirely QUANTIZATION noise.  In fact, because it is a 1-bit system, it CANNOT be fully dithered.  Which means YES, ultrasonic noise, which is noise-shaped quantization noise from the 1-bit samples, is correlated to the audible range.  Dither is random noise than de-correlates quantization noise in mulit-bit PCM systems.  It isn't something that can be accomplished, at least not fully, in 1 bit systems.  

That is the other thing the dude told me, that the DSD dither noise is not correlated at all to any harmonics in the audible band.   I don't know where people go so wrong on something so very, very basic.  (I warned you I would not be very tactful about this experience.  Sorry if you are offended, but you don't have to read lol.)

Then he asked me if I knew that most DSD was actually edited in PCM.  Again, these 'ASS'umers.  Of course I know that.  Of course I also know there is a fairly large for a niche market 'PURE' DSD industry that uses minimal DXD punch-in/punch outs, crossfades etc, but the majority is made to stay in DSD.  Also, there was this thing called DSD-wide, that is a totally different story for another day, but it also allowed the same kind of minimal editing.  You didn't have to convert everything in its entirely to multi-bit.  And even if the system is converted to multi-bit, it isn't exactly a bad thing.  DSD's advantages, if it has any, are not defined so much by its bit-depth as it is the sample rate, and the filtering. (Which is why the original DSD should have at least been a few levels, rather than just 1-bit.)  Even most 'Pure DSD' DACs convert 1-bit DSD into multiple levels of that 1-bit signal, offset in time by a single clock sample, to filter it.  This can be done in a totally digital form, with taps that multiply every stream (anywhere from 4 to 32 stacked streams are what I have found) by 1, meaning the same comes out as went in, and all the filtering is done in the 'delay', actually making this FIR filter as much as CIC filter as anything, with no decimation stage.  Or it can be done almost exactly the same way, except the filter can be implemented at the output stage itself, with the resistor/switch being the TAP, filtering the multiple streams of DSD AND converting them to analog at the exact same time.  Pretty efficient and ingenious.  

Anyway, no! DSD is not dither noise.  I think people get this idea from the most basic of explanations that use black and white pictures.  If you have a 1-bit pixelated black and white video system, and try and draw an image, you will get completely black shapes, with maybe a recognizable outline, against an all white background.  If you randomize the noise instead, sending some white pixels into the black, and some black pixels into the white, all of a sudden the eyes can see a more detailed image, albeit with a 'haze' of noise uniformly across it.  I have seen this used to describe how DSD works.  

But it actually is nothing like how DSD works.  This is indeed a good description of dither.  And maybe on some very simple conceptual level it is helpful in beginning to understand DSD or 1-bit systems.   But again, this is ultimately wrong when it comes to audio, quantization noise, DSD and Noise Shaping.  

Finally, this 'educator' attempted to put down any notions of psychoacoustics playing a role in the sound of various formats like DSD.  Of course, be brought no references.  Or perhaps he works like Gemini AI and uses inaccurate forum threads (Gemini used more than just the one I posted on, almost all its references are from user run audio 'science' forums).  Let's finish this up with exactly what I was talking about before he rudely 'ass'umed I was a village idiot.  (I'm not the village idiot.  I am more like the guy who is smart enough to count out the dinari at the market and make sure no one is stealing.  So no, I am not the smartest guy by any means, but I am not the dumb one either.) 

Psychoacoustic research into why some listeners perceive DSD (Direct Stream Digital) as sounding better than other digital audio formats, such as PCM (Pulse Code Modulation), involves exploring how humans perceive sound and how different audio encoding techniques interact with our auditory system. Here are several factors that contribute to the perceived superiority of DSD:

Key Factors in Psychoacoustics and DSD Perception

High Sampling Rate:

DSD Sampling Rate: DSD uses a very high sampling rate of 2.8224 MHz (64 times the CD standard of 44.1 kHz). This high sampling rate can capture more of the audio spectrum, leading to a perception of more natural and dynamic sound.

Psychoacoustic Impact: Humans are sensitive to high-frequency content transients. The high sampling rate of DSD may better capture these transient elements, due to the ability to capture faster transients, and the potential lack PCM type filtering artifacts, dependent on filter parameters that take advantage of DSD benefits, enhancing the perception of realism and presence in the audio.

Noise Shaping:

Quantization Noise: DSD uses noise shaping to push quantization noise to higher frequencies, well beyond the range of human hearing (20 Hz to 20 kHz). This means the audible band is relatively free of quantization noise.  

Psychoacoustic Impact: A lower noise floor in the audible range can lead to a cleaner and more transparent sound. Listeners might perceive the audio as having more depth and clarity.  It is true that very high bit depth PCM also has low quantization noise, however, all the quantization noise power, even if low in level stays in a much more narrow range, much of it the audible range, almost all of it in the audible range if the sample rate is 44.1khz.  For PCM the uniform distribution of quantization noise could still affect the subtle nuances of the audio.  By shifting noise to the ultrasonic range, DSD may preserve more of the delicate details and spatial cues within the music, enhancing the perceived realism and depth of the audio.

One-Bit Signal Processing:

Simplicity: DSD uses a 1-bit signal, which some argue leads to less complex processing and potentially fewer artifacts compared to multi-bit PCM.  This is especially so the less DSP is required, and the fewer modulations before conversion.  

Psychoacoustic Impact: The simplicity of the 1-bit signal may result in a more coherent and phase-accurate reproduction, which can enhance the perception of spatial accuracy and instrument separation.

Subjective Preference and Listening Environment:

Individual Differences: People have different auditory sensitivities and preferences. Some listeners might be more attuned to the qualities that DSD enhances, such as high-frequency detail and low noise.

Listening Environment: High-quality playback equipment and acoustically treated listening environments can make the differences between DSD and other formats more noticeable.

Research and Studies:
Several studies and research papers have explored the subjective perception of audio quality between DSD and PCM. Some key findings include:

Listener Preference: Controlled listening tests have shown that some listeners prefer DSD over PCM, citing smoother and more natural sound.

Critical Listening: Trained listeners and audio professionals often report differences more accurately, suggesting that experience and familiarity with high-quality sound influence the perception of DSD.


Psychoacoustic Advantages of Ultrasonic Harmonic Noise in DSD

In DSD, ultrasonic noise is typically harmonically related to the audio signal due to the nature of delta-sigma modulation. This harmonic structure can extend well beyond the human hearing range (20 Hz to 20 kHz).

Perceived Sound Quality:

Subharmonic Effects: Although the ultrasonic frequencies are above the audible range, their harmonic relationships can influence subharmonic frequencies within the audible range through intermodulation distortion, which can enhance the perception of a richer and more complex sound, even sometimes at the expense of measured performance.

Inaudible Frequencies: These frequencies might interact with the auditory system in ways that affect the perception of lower frequencies, potentially adding to the sense of depth and spatiality in the audio.

Localization Cues: Ultrasonic frequencies can influence spatial localization cues, potentially enhancing the perception of the soundstage. The brain processes these cues to determine the location of sound sources.

Ambience and Air: The presence of ultrasonic harmonics can contribute to the perception of ambience and airiness in recordings, leading to a more lifelike and immersive listening experience.

Influence on Lower Frequencies:

Nonlinearities in Hearing: The human auditory system exhibits nonlinearities, meaning that interactions between ultrasonic frequencies and audible frequencies can generate audible artifacts or enhance existing tones.

Masking Effects: Ultrasonic content can create masking effects, altering how lower frequencies are perceived. This can lead to a cleaner and more detailed perception of the mid and low frequencies.

Subjective Preference for all High Resolution formats:

Listener Preference: Many listeners subjectively prefer audio with rich harmonic content, including ultrasonic harmonics, as they may contribute to a perception of higher fidelity and naturalness.

High-Resolution Audio: Audiophiles often report that high-resolution audio formats (like DSD) that include ultrasonic content sound more realistic and engaging compared to standard-resolution formats.

Conclusion:
The perceived superiority of DSD to some listeners can be attributed to its high sampling rate, effective noise shaping, and the psychoacoustic impacts of these factors. The subjective nature of audio perception means that individual preferences and sensitivities play a significant role in how DSD is experienced compared to other digital audio formats.



References and Studies: (the most important part)

Psychoacoustics: Facts and Models by Hugo Fastl and Eberhard Zwicker: Comprehensive coverage of how the human auditory system processes complex sounds, including the effects of ultrasonic frequencies.

The Influence of High-Frequency Audio Content on the Perception of High-Resolution Audio: This AES convention paper investigates how high-frequency content influences the perceived quality of high-resolution audio.

Intermodulation Distortion in Digital Audio Converters: Discusses how ultrasonic frequencies can create intermodulation products that fall within the audible range, potentially enhancing the richness of the sound.

The Effect of Ultrasonic Components on the Perception of Music: A study examining how ultrasonic components in music recordings affect listener preferences and perceived audio quality.

Perceptual Audio Coders: What To Listen For by James D. Johnston: Offers insights into how various audio coding techniques and their handling of ultrasonic content can affect perceived audio quality.

"The Perception of High-Frequency Content in Music": This paper discusses how high-frequency content affects perceived audio quality.

AES Journal Articles: The Journal of the Audio Engineering Society has published numerous articles on the psychoacoustics of digital audio formats, including DSD and PCM comparisons.

I was thinking one day I need a super cheap portable DAC for another baseline reference device in my reviews.  Not necessarily baseline measurements; it isn't difficult to make a DAC measure well these days.  I was thinking actual sound quality and how cheap could a device be before it was no longer enjoyable.  

I also wanted something with an actual Direct DSD path.  So ESS was out.  That really meant AKM, or Burr-Brown, and I already have plenty of iFi products around with the Burr-Brown DSD1793, so I chose an AKM product because I previously had a great experience with the AKM4493 in the RME ADI-2 PRO.    The AKM chip isn't quite as DIRECT DSD ala Signalyst or similar, that keep the DSD signal at 1-bit all the way to the FIR filter that converts DSD to analog.  In the Signalyst DAC, the filter itself becomes the digital to analog converter with shift registers, resistors and switches.  (What COULD have been the truest, most direct DSD DAC ever brought to market was the PSAudio Directstream because its filter is purely analog, not a digital filter implemented by analog components or some combo thereof. Unfortunately, like the ESS chipset, there is no way to bypass the quite massive DSP applied to both PCM and DSD formats as they enter the Directstream.)

The AKM chips with Switched Capacitor Filters are really, really good chips.  Then AKM had their terrible factory fire, and the newest chips are now outsourced and have moved away from SCF's to resistor based elements like the Signalyst, Burr-Brown, ESS, well, like a LOT.  It changes a LOT of things and I have seen lots of confusion in otherwise professional reviews on how DSD works in AKM based devices.  

Here is a quick rundown on how it works with the SCF chips like the 4493 (and presumably still kind of the same with their new resistor-based chips, but not quite the same as the other resistor-based chips from other brands.)  

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From the block diagram of the AK4493 DAC, it is evident how the DSD data is processed in bypass mode and normal mode. Here's a detailed explanation of the volume control and delta-sigma modulation in the AK4493 DAC, based on the provided information and the datasheet.

Bypass Mode for DSD (DSDD1)

1. DSD Data Interface & Filter:
   • The DSD data interface and filter block handles the incoming DSD data. When the DSDD bit is set to '1', it indicates the bypass mode.
2. FIR Filtering:
   • In bypass mode, using AKM logic code DSDD1, the DSD signal is filtered using a FULLY DIGITAL Finite Impulse Response (FIR) filter. As mentioned earlier, many Direct DSD DACs don't use a fully digital FIR filter.  They use digital principles but are implemented with analog components, meaning the TAPS and the analog switches are the same thing.  In the AKM 4493 however, this digital filtering stage is implemented much earlier in the logic process, meaning the logic is fully digital, with digital TAPS all equally weighted at a value of 1.  Slightly different means which ultimately lead to the same end, ensuring that the high-frequency noise inherent in DSD signals is attenuated.  The fully digital filter outputs a multi-bit signal at the same sample rate as received.  There is NO further noise modulation, as step three will elaborate further.  This signal will be in unary code, and it could stay that way during its transmission to the Switched Capacitor Filters, or it could be immediately converted by digital logic to binary and then re-converted to unary code at a later stage before the Switched Capacitor Filters, which I believe is the most likely scenario.
   • Skipping ΔΣ Modulation:
     • The key aspect of bypass mode is that it skips the delta-sigma (ΔΣ) modulator. Normally, the ΔΣ modulator would convert the filtered DSD signal into another high-frequency pulse-density modulated signal with different characteristics.  However, in bypass mode, this step is omitted to preserve the original DSD signal characteristics as much as possible.  This is ,after all, pure DSD.  Or as about as pure as it gets.  (It has to be filtered somewhere, that cannot be avoided.  As long as it goes through no more DSP, it doesn't really matter if the filter is at the beginning of the chain or the end.)
3. Unary Code Output:
   • The filtered DSD signal is either in unary code or converted to unary code before it reaches the switched capacitor filters (SCFs). Unary coding is beneficial for reducing digital switching noise and improving linearity in the final conversion stages by allowing scramble code/dynamic element matching.  (It is THE standard for Delta Sigma DACS of any N-bit design.) 
4. Switched Capacitor Filters (SCFs):
   • The unary coded signal is fed directly into the SCFs, which perform the final digital-to-analog conversion. The SCFs average the high-frequency pulses to produce a smooth analog signal, effectively filtering out high-frequency noise and yielding a clean analog output.  This combined with the earlier non-decimating FIR filter with equally weighted taps create a very powerful tool for shaping the DSD signal.  Perhaps the most powerful on chip you will find.  

(Quick note for below... we are now describing a different process, how DSD is converted when the Bypass mode in NOT used, just in case there is any confusion.)

DSD Processing in Normal Mode (DSDD0):

1. DATT with NO Attenuation 
   • LOCKING the DATT volume control to 100% ensures that the signal's amplitude is not altered. This setting is equivalent to bypassing the volume control but allows the signal to pass through the volume control logic unaltered.  THE SAME WILL APPLY FOR PCM IN THIS MODE.  THE AKM DIGITAL LOGIC IS USED IN THIS WAY TO ACHEIVE FIXED OUTPUT MODE FOR BOTH SIGNALS.  ALSO NOTE THAT SIMPLY LOCKING THE VOLUME AT 100 PERCENT DOES NOT SWITCH THE SYSTEM TO LOGIC DSDD1 FOR BYPASS MODE!!!  THIS IS THE MISTAKE I HAVE SEEN MANY PROFESSIONAL REVIEWERS MAKE WITH AKM CHIPSETS! 
2. DATT WITH VOLUME CONTROL/ DSD Attenuation
   • The incoming DSD signal is received and initially processed by the DSD Data Interface and Filter. This block includes the necessary aforementioned FIR filtering.  The filter, either 1 or 2, in this mode is to manage high-frequency noise inherent in DSD signals, AND just as importantly in this case, to create a multi-bit signal that will be manipulable by a volume/gain control if and when needed.
   • The DSD signal then passes through the Digital Attenuation (DATT) block. Here, the volume control is applied to the DSD signal. This block allows for precise digital volume control, attenuating the signal as required. This step is crucial when volume control is desired for DSD playback.  The output of the 'Normal' path DSD filter was almost certainly converted into a binary code exactly equivalent in value to the unary code produced by the FIR filter with equally weighted taps.  This is because the binary code will allow for much more precise volume control and actually will require less overhead to work.  

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 ​FURTHER PROCESSING OF DSD NORMAL MODE

1. DSD Data Processing:
   • When the DSDD bit is set to '0' (normal mode), the DSD signal does not bypass the ΔΣ modulator. Instead, the DSD signal undergoes the standard processing path, which includes noise shaping and modulation by the ΔΣ modulator.
2. Delta-Sigma Modulator (ΔΣ Modulator):
   • The ΔΣ modulator remodulates the filtered DSD signal into a multi-bit Delta Sigma signal.  This process helps shape the noise, pushing it out of the audible frequency range and improving the overall signal quality, potentially more-so than a single bit DSM system, as a multi-bit DSM system has much less quantization error to shape.  
3. Switched Capacitor Filter (SCF):
   • The modulated signal, if not already in unary code is converted to unary code, undergoes Dynamic Element Matching/code scrambling, and is then fed into the SCF, which performs the final digital-to-analog conversion. The SCF averages the high-frequency pulses to produce a smooth analog output, effectively filtering out the high-frequency noise.

FURTHER PROCESSING OF DSD BYPASS MODE

1. DSD Data Processing:
   • When the DSDD bit is set to '1' (bypass mode), the DSD signal bypasses the ΔΣ modulator.  It has already been a delta-sigma signal once before, nor has it been touched by any digital volume control or a redundant remodulation as has the DSD bitstream in normal mode described above.  Also, it has already been filtered by a digital FIR filter without decimation, so it is simply ready to be converted directly into analog.  
2. Switched Capacitor Filter (SCF): 

The filtered oversampled multi-level DSD signal is then fed into the SCF, which performs the final digital-to-analog conversion. The SCF further filters the high-frequency pulses to produce and even smoother final DSD signal, and it the process concerts the signal from digital into analog.  


SOME CONCLUSIONS

Benefits of Using the Normal Path for DSD with Volume Control:

• Consistent Volume Control: Applying digital volume control to DSD signals allows for consistent attenuation across both PCM and DSD formats.
• Enhanced Noise Shaping: By passing the pre-filtered DSD signal through the ΔΣ modulator, the DAC can effectively reshape the quantization noise, pushing it further out of the audible range and improving audio quality.  Since the remodulation is done with a multi-bit quantizer, this allows for greater consistency between DSD speed formats and is much better at handling the ultra-sonic quantization noise.  
• Flexibility: This setup provides flexibility in managing volume levels digitally while maintaining high fidelity in the analog output.

By using the normal path (DSDD = 0), the AK4493 DAC can apply volume control to DSD signals, ensuring that users have the flexibility to adjust playback levels while benefiting from the advanced noise shaping and modulation techniques integrated into the DAC.

Benefits of Using Bypass Mode of Volume Control and Modulator:

1. Preservation of DSD Characteristics:
   • Bypass mode allows the DSD signal to maintain its original 1-bit, high-frequency characteristics, dependent on the quality and parameters of the pre-filtering. This can be important for purists who prefer the unique sound quality and characteristics of DSD audio, which can be altered by further digital processing.
2. Reduced Processing Complexity:
   • By bypassing the ΔΣ modulator, the signal processing path is simplified. This reduction in processing stages can result in lower latency and fewer opportunities for digital artifacts to be introduced into the signal.
3. Lower Power Consumption:
   • Skipping the ΔΣ modulation stage can reduce the overall power consumption of the DAC. This is beneficial for battery-powered devices or applications where power efficiency is critical.
4. Direct Digital-to-Analog Conversion:
   • The DSD signal, after FIR filtering, is converted directly to analog using the Switched Capacitor Filter (SCF). This direct path can result in a cleaner and more transparent signal path, which some audiophiles may prefer.
5. Simplified Signal Path:
   • A simpler signal path with fewer stages can enhance the overall reliability and stability of the DAC operation. Fewer processing stages mean there is less chance for signal degradation or synchronization issues.
6. High-Fidelity Playback:
   • For high-resolution audio playback, preserving the integrity of the original DSD signal can yield a more accurate and high-fidelity sound. This can be particularly noticeable in high-end audio systems where every detail of the audio signal is critical.

Use Cases for Bypass Mode:

• Audiophile-Grade Audio Equipment: High-end DACs used in audiophile-grade audio equipment often prioritize maintaining the purity of the original audio signal. Bypass mode is ideal in these scenarios.
• Battery-Powered Devices: Portable audio devices that rely on battery power can benefit from the reduced power consumption in bypass mode.
• Minimalist Design Approaches: Audio systems designed with a minimalist philosophy, aiming to use the least amount of processing possible, can leverage bypass mode to achieve their design goals.

Conclusion:
​Bypass mode in the AK4493 DAC offers a streamlined and purist approach to digital-to-analog conversion for DSD signals. It preserves the original characteristics of the DSD signal, reduces processing complexity, lowers power consumption, and provides a simplified signal path that can be beneficial in high-fidelity audio applications. This mode is particularly suitable for audiophile-grade equipment where maintaining signal purity is paramount.

SO THEN, ANDREW, WHAT WAS THE BIG DEAL? WHY DID YOU CALL OUT SOME ONLINE AND PAPER MAGAZINES FOR SAYING THAT A DIFFERENT TOPPING PRODUCT (THE E70V), WITH A TOTALLY DIFFERENT AND NEW AKM CHIP,  INDEED ALLOWED ACCESS TO THE PURE DSD BYPASS MODE?  (WHEN IT OBVIOUSLY DOES NOT.)

We will leave aside the fact that it's a totally different chip for later, but getting ahold of this Topping E30 II Lite has shed a bit of light on the 'controversy'.  You see, the advertising propaganda for the E30 indeed says that it offers the true DSD BYPASS mode.  It instructs its users to simply put it in FIXED OUTPUT mode, and the DSD will not have a volume control and therfore will bypass the internal modulator.  This is both stated and implied.

In the case of the Topping E30 II Lite, that MAY indeed be the case.  I have spent hours cooking up multiple tests to sniff out the truth, but the hard facts are, no matter in Fixed or Variable Output, everything measures EXACTLY the same!!!  And knowing how direct bitstream DSD interacts with analog output stages in a very different way than non-direct DSD that take full advantage of the performance gains offered by multi-bit Delta-Sigma noise shaping, my Spidey Sense is up.  But I can't find as of yet a true smoking gun with this particular AK4493 chip in this particular Topping E30 II Lite DAC.  The jury is still out, but my opinion is that NO, it doesn't use the bypass mode at all when you lock the volume control at 100 percent (no attenuation on either DSD or PCM).  

That brings me around to the products I reviewed with the latest AKM dual chip AK4191 + AK4499.  I had my first experience with this very different AKM chip in the Topping E70V Velvet. The controversial one. The thing about this chip or chips, is they are VERY different from the more well known and highly regarded AK4490, AK4493, etc, which were all based around switched capacitor conversion, and AKM were the MASTERS at it.  Then comes that dreadful factory fire, and things really changed.  Not only were a lot of our chips now being outsourced, AKM switched (no pun intended) from what they do best in Switched Capacitors over to Switched Resistors.  Really, this is a whole new ball game.  

And now for a little speculation.... in the past perhaps it was a common practice when using the AK449x chips to activate the DSD bypass mode when also 'deactivating' the Volume control for full fixed output across formats.  Makes total sense.  But this has to be programmed in the chip logic to happen that way.  It is two different actions.  And they absolutely do NOT have to be performed at the same time.  When I reviewed the Topping E70V Velvet, I got the same Spidey senses I mentioned with this Topping E30 II lite.  The two modes, volume control on, and volume control fixed or 'bypassed' measured exactly the same.  Once again, not a thing in the measurements to suggest this had two different paths for DSD conversion.  It certainly still could have been the case, so I messaged Topping directly and they directly got back to me and said in no uncertain terms that 'NO', the E70V does not offer the bypass mode.  

And for more confirmation, the other product I have reviewed with the AK4191 + AK4499 chipset, the SMSL D400, actually has a THIRD entry under the menu that specifically has a selection for 'DSD BYPASS MODE', along with the other two modes, that simply determine whether the DAC is used as a pre-amp with volume control, or as a DAC only with fixed volume output.  You want BYPASS MODE DIRECT DSD?  No other way to do it except to select that particular, unambiguous option.  Just selecting to use fixed volume control will not cut it.  

And remember, the technology in THIS multi-chip AKM 4191 + 4499 DAC is totally different than previous AKM DACs, and a 'deep', well not so deep dive into the SMSL version's measurements shows massive differences in the filter behavior and overall performance characteristics that I was fully expecting to see in a DAC that actually has two different DSD modes available to activate.  So, there is NO DOUBT about those two DACs.  The Topping E70V?  NO PURE DSD BYPASS.  The SMSL D400?  YES, YES, YES it has the PURE DSD BYPASS OPTION.  (And did I mention this was entirely new tech for AKM that differs pretty massively from their bread and butter?  Yeah, it needs some firmware work and let's leave it at that.)

But this little Topping E30 II Lite?  I am 90 percent sure it does NOT allow access to the DSD Bypass mode in spite of advertising it prominently as a feature.  Surely no  company has ever gotten something wrong, exaggerated, or just flat out lied? 

And as I have thought about it some more, considering this is a super small, super cheap device that costs less than most 2 meter RCA interconnects these days, why even SHOULD it have the extra logic programming to do something it doesn't need to do?

Because it measures admirably well in both PCM and DSD modes, both fixed output and variable volume output.  DSD measures identically in either output mode.  And the actual filtering they are using on DSD is EXTREMELY gentle, which allows one of the biggest strengths of DSD to shine out, and that is the transient response.  Also, it allows enough ultrasonic noise to enter the ears, and even though we cannot hear it, that ultrasonic quantization noise, unlike random PCM quantization noise, stays harmonically related to what we can hear.  Psycho-acoustic experts theorize that this plays a big part in why DSD sounds so 'good' to many people.  It goes beyond our basic hearing and how the noises are processed in our neural networks.  And that is where all the REAL work is done!  Between the ears!  And, well, with the ears too.  This blog entry has gotten way to long already so I will save more info on why DSD could sound better for another day.   

And finally, I am back to pondering the fact that this is a cheap product in which there is no way it has the ability to articulate the minute differences that might exist between a pure DSD bypass mode conversion and one that decides to not take the bypass, yet would rather taxi right on into the Modulator City.  

I will have a more proper review soon, locatable under the 'review' tab you see above.  It won't go over all this stuff again; I will just link to it where appropriate.  But now a preview of the review.. The Topping E30 II Lite is a good sounding little product for the price, and measures way too good for the price.  See you on the other side of that review!

Multitone vs REW.  Two extremely important and wonderful free software programs in the audio measurement sphere.  I am not sure which qualifies as the Swiss Army Knife, and which as the Scalpel.  They both have their strengths and weaknesses, but for basic measurement tasks they produce very similar results within any reasonable margin of error.  

I ordered a Topping E30 II Lite DAC (DAC only, no headamp), which is about as cheap a 'good' DAC as your money can buy.  I am putting it through its paces to create an actual 'lower-end' reference to refer all my DAC/Pre-amp reviews back to, however, I continue to be impressed by the measured performance of even the cheapest Chi-Fi products.  

I cannot make any reliability claims, though.  I have very little time with the DUT.  Also, I cannot blindly claim "it measures so well it MUST sound as good or better than more expensive products!"  That is a great way to placate oneself as and end-user when you cannot afford better.  The fact remains there are nice measuring products that sound as well as my Yorkie's crap stinks, while there are 'good enough' measuring products that may fall short in the technical camp according to some, yet sound truly 'audiophile'.

Back to the point of this post entry; what does each measurement suite say about the Topping E30 II Lite DAC.  They don't give out exact results, but IMO the results are well within any margin of error and won't contribute to any audible issues. 

These measurements were taken with only the E1DA COSMOS ADC in play.  The real special sauce that allows these programs to go from fairly accurate to downright giant killers is the E1DA COSMOS APU external notch filter.  That was not used here; this is just a quick look-in at two different programs measuring the same DUT under the exact same conditions and equal parameters.  They stack up well against each other.  

​

THD w/Multitone = -120.2db
THD w/Room EQW= -120.3db

THD+N w/Multitone = -104.6db
THD+N w/Room EQW= -104.db

This is just the tip of the iceberg, as they say.  I have a backlog of measurements to make, and my hardware (and knowledge how to use it) keeps improving week to week, thanks to an extremely generous benefactor that has me contemplating an Audio Precision test unit.  As you will see as we progress through the Blog section of EuphonicReview.com, the E1DA suite of tools combined with the available readily attainable software is so good, it may be in the best interests of Euphonic Review to pocket any money earmarked for the 20 grand at minimum Audio Precision.  

After all, the next major addition to the website is tube reviews and sales, and I have already invested a hefty sum into an Amplitrex AT1000 tester.  

It is a golden era for testing all manner of devices.  I hope you are enjoying or will enjoy this common journey at which ends audio nirvana.  

​

This is the third and final 'dive' into some of my best and favorite tubes in my collection.  We have seen the usual suspects, such as RCA Black Plates and Telefunken Smooth Plates, but thrown in to the mix have been some rarely known Japanese tubes, and some French tube relationships that you may not have know about.... oh those French!  Today's entry contains a similar eclectic mix.  I hope you enjoy!
​

Remember when I noted some people think Japanese tubes are bunk?  Poppycock.  Does anyone think Western Electric is bunk?  Hmmm.   I didn't think so.  

NEC (Nippon Electric Company) Japan has a significant history in the production of vacuum tubes, a journey that began in the early 20th century. The company, established in 1899, entered the vacuum tube industry with the aim of supporting Japan's growing telecommunications needs.

NEC's venture into vacuum tubes was greatly influenced by Western Electric, the manufacturing arm of AT&T. In the 1920s, Western Electric provided NEC with critical technology and expertise, enabling the Japanese company to produce vacuum tubes domestically. This partnership was part of a broader strategy by Western Electric to expand its influence and ensure a reliable supply chain for its telecommunications infrastructure worldwide.

With Western Electric's support, NEC rapidly advanced its manufacturing capabilities. By the 1930s, NEC was producing a wide range of vacuum tubes, including those used in radios and early television sets. The company's tubes were known for their reliability and performance, helping to establish NEC as a leading electronics manufacturer in Japan.
​
During World War II, NEC's production shifted to support the war effort, producing tubes for military communications and radar equipment. This period saw significant technological advancements and the expansion of NEC's manufacturing facilities.

After World War II, NEC resumed its focus on consumer electronics and telecommunications. The company continued to innovate, developing new types of vacuum tubes that were essential for the rapidly growing electronics market. In the 1950s and 1960s, NEC's vacuum tubes were widely used in televisions, radios, and early computers, solidifying its reputation as a pioneer in the industry.

And some of the very best sounding tubes in the WORLD are those 1950's and 1960's Nippon tubes, especially the long plate 12AX7's.  I would place them on the same playing field as Telefunken.  They really have a strong sonic resemblance to the clean, clear, very detailed Telefunken sound, also having a very similar touch of mid-range warmth. 

If Telefunken is the starring tube of the West, then NEC is in my opinion the starring tube of the East.  

The NEC long plate 12AX7 from the early 1960's is a gem cherished by audiophiles and vintage equipment enthusiasts alike. 
What sets the NEC long plate 12AX7 apart is its unmatched reliability and consistency. Manufactured with meticulous attention to detail, these tubes exhibit minimal microphonics and maintain their performance over extended periods of use. Whether installed in a vintage guitar amplifier or a state-of-the-art preamp, the NEC 12AX7 provides an unparalleled audio experience that few modern tubes can replicate. Its enduring popularity is a testament to NEC's engineering excellence and the timeless appeal of these classic vacuum tubes, making them a prized component in any audiophile's collection.

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)