Preface: DSD Is Not Magic — It's Finer Noise Management


In the Hi-Res audio camp, DSD (Direct Stream Digital) has always been synonymous with "analog flavor." Since Sony and Philips jointly proposed the SACD specification in 1996, debate around DSD has never stopped: some revere it as the ultimate form of digital audio, others argue it's fundamentally no different from PCM — indistinguishable even in double-blind tests.


Today we need a more rigorous perspective.


What always determines sound quality first is recording quality and environmental noise floor — no format can remove the noise in an original recording. The core technology of PCM-to-DSD conversion is essentially using noise shaping and upsampling to "push" quantization noise into the ultrasonic band where the ear is insensitive, thereby improving in-band SNR and making the sound cleaner and more natural — but the noise floor of the original recording remains, a physical reality every digital format faces.


Based on this understanding, this article dissects DSD's true value at three levels:


  1. Technical principles — how noise shaping and upsampling work
  2. Vendor implementation — differences in SDM, upsampling and DAC design across brands
  3. Playback experience — how these differences ultimately shape what we hear


Chapter 1: From PCM to DSD — Not Eliminating Noise, But Managing It


1.1 PCM's Limitations


PCM (Pulse Code Modulation) is the standard digital audio format of the CD era, recording amplitude values at fixed sample rates with multi-bit depth (16-bit, 24-bit). PCM's quantization noise is uniformly distributed across the band from 0 Hz to Fs/2 (the Nyquist frequency).


For CD spec (44.1kHz/16bit), quantization noise is uniformly distributed across 0–22.05kHz — exactly covering the entire human audible band (20Hz–20kHz). This means CD-spec PCM exposes its quantization noise fully within the sensitive hearing region.


Raising bit depth (e.g., 24-bit) lowers the quantization noise level, but the noise remains uniformly distributed in-band — it cannot fundamentally change the situation of "noise coexisting with signal at the same frequencies."


1.2 DSD's Technical Approach: Noise Shaping + Upsampling


DSD takes a fundamentally different approach. It abandons multi-bit amplitude recording in favor of a 1-bit bitstream at extremely high sample rates (2.8224MHz — 64× CD), recording the *direction of change* (up or down) relative to the previous moment, not absolute values.


When we upsample PCM to DSD, two key events occur:


First, upsampling and noise spreading. After the sample rate jumps, the total quantization noise energy is "flattened" across a much wider band. Compared to 22.05kHz of bandwidth, 2.8224MHz is 128× wider, so noise density per unit bandwidth drops to about 1/128. In other words, the in-band noise floor becomes theoretically quieter — not by eliminating noise, but by spreading equal-energy noise across far more space.


Second, noise shaping (Σ-Δ modulation). This is the step where DSD truly unlocks its potential. Through a high-order Σ-Δ modulator, most quantization noise is pushed into the ultrasonic region above 20kHz (e.g., 30–100kHz), while the human hearing limit is about 20kHz. Thus the SNR within the audible band (20Hz–20kHz) is greatly improved and the signal purity is higher.


Think of it this way: PCM's quantization noise is like a basin of water spilled on the floor, evenly covering every corner of the audible range; DSD's noise shaping is like channeling the water into a drain in the corner of the room, keeping the main activity area relatively dry — the water is still there, just not where you actually are.


1.3 The Boundary That Must Be Clear


But please be clear: this processing only reshapes quantization noise; it cannot erase the environmental noise floor captured by microphones or the device's own electrical noise — those are source information, and no format conversion can touch them. DSD's wisdom lies in not polluting the original information, only optimizing its own quantization pollution, so the original recording's noise floor is no longer stacked with extra quantization noise, presenting the original sound background more faithfully.


In other words, DSD is a "purifier," not a "repairman." It can enhance a good recording but cannot resurrect a bad one.


1.4 The Progress of an Era: Compute Power Opens New Space for SDM


DSD's technical principles were complete by the early 1990s, but between theoretical feasibility and practical usability stood a compute-power chasm.


The reality when the SACD spec was introduced in 1996:


When Sony and Philips launched SACD in 1996, the selling point was precisely "closer to analog sound, more faithful." But the reality of the 1990s: consumer DSP compute was extremely limited, FPGAs were not yet widespread, and high-performance Σ-Δ modulators depended on dedicated ASICs. Constrained by that era's compute ceiling, SACD's SDM parameters had to be conservative — low modulator order, shallow noise-shaping curves, limited arithmetic precision.


The result: DSD64's (2.8224MHz) quantization noise was pushed outside the audible band, but not very far. Under the earliest SACD spec, quantization noise still had clearly visible residue around 20–25kHz, with very little margin from the perfect human hearing limit (~20kHz). This directly led to a strategy: when quantization noise can't be pushed far at the same multiplier, the simplest solution is raising the sample multiplier — from DSD64 to DSD128 to DSD256, doubling the rate to give noise shaping more frequency headroom and push audible-band quantization residue further away.


This is the first era in DSD history: fixed noise-shaping parameters, relying on stacking DSD rates for better noise performance.


1.5 The Compute Revolution Inflection Point


From the mid-2010s, the situation changed fundamentally. FPGA performance soared, DSP processors entered the multi-core era, and GPU computing entered audio processing. These qualitative compute changes finally gave SDM designers ample room to work.


Today's top SDM modules (e.g., HQPlayer's ASDM5/ASDM7, DpdoEngine's noise-and-harmonic-optimized SDM) achieve: at the same DSD64 rate, through more complex, higher-order modulator design, push quantization noise from the old ~25kHz level to 35kHz or even 50kHz+.


What does this mean?


  • Noise-shaping performance that once required DSD128 or even DSD256 can now be achieved at DSD64 — higher rates still have advantages, but the gap has shrunk dramatically.
  • SDM algorithm design is no longer constrained by compute; it can attempt more complex noise-shaping curves (e.g., dual-band NTF with frequency-weighted priorities), finer dither strategies, even machine-learning-based adaptive modulation.
  • Real-time PCM→DSD conversion becomes possible — players can perform high-quality upsampling and noise shaping live during playback, without relying on pre-conversion or hardware ASICs.

If the first era's slogan was "stack rates," the second era's slogan is "optimize algorithms."


This shift has profound industry impact. Previously, DSD sound quality depended almost entirely on hardware DAC design; now, an excellent software SDM plus a DAC with DSD Direct capability can rival or surpass many traditional FPGA hard-decode solutions. This is why more high-end streamers put conversion compute on the front end (computer or streamer core), letting the DAC focus purely on D/A conversion.


More importantly, the compute leap broke the "DSD is good but high-barrier" dilemma. A mid-range device with excellent software SDM can deliver DSD playback quality that may exceed a decade-old flagship relying purely on rate stacking. DSD is no longer a luxury for the few, but an audio optimization technology more people can enjoy.



Chapter 2: Noise and Time — DSD's Real Value in Listening


Since DSD can't change recording noise floor, what do we actually hear differently in playback? The differences show up in three areas:


2.1 A "Blacker" Audible Band


After noise shaping, in-band noise power drops dramatically; the background feels pitch-black and quiet, and micro-details surface more easily. This is why many audiophiles report "clearer instrument overtones and richer low-level signals" with DSD. DSD didn't "create" new detail — it removed the in-band quantization noise PCM leaves behind, letting faint signals that were recorded but masked by noise emerge.


An analogy: the same photo, the film itself unchanged — but when a layer of dust on the glass plate is wiped away, the previously blurred details naturally become clear.


2.2 More Natural Transients


DSD's 1-bit pulse-density modulation inherently has extremely low phase error. PCM's multi-bit decoding suffers from inherent "zero-crossing distortion" — when a signal crosses from positive to negative half-cycle, multiple bits of the multi-bit decoder switch simultaneously, producing nonlinear distortion. Additionally, PCM needs steep anti-aliasing filters to rebuild analog waveforms, which triggers the Gibbs phenomenon (ringing) at sharp signal changes.


DSD's Σ-Δ modulation is essentially a continuous 1-bit pulse stream — no multi-bit switching, hence no zero-crossing distortion. Its impulse response is more linear in the time domain, especially in the 5kHz–10kHz mid-high band, where attack and decay sound more natural. Audibly: string friction feels more real, vocal sibilance is softer, cymbal overtones have more air.


2.3 Less Listening Fatigue Over Long Sessions


Because high-frequency quantization noise is pushed into the ultrasonic region, there's no sharp filter ringing in the audible range; the overall timbre is rounder and warmer, evoking the "flow" of analog tape. This difference is especially obvious comparing the same album's PCM 24bit/96kHz version against DSD64: the DSD version often has a more open soundstage depth and more stable imaging, yet an identical noise floor — because the noise floor comes from the master itself.


This isn't just subjective. Research shows the ear unconsciously fatigues from high-frequency noise and distortion in the audible band — like feeling tense in a long noisy environment. By pushing noise out of the audible range, DSD objectively reduces the auditory system's burden, allowing longer focused listening.



Chapter 3: SDM — DSD's "Soul," But Every Maker Writes It Differently


Going deeper to the core question: DSD is ultimately a storage carrier for Σ-Δ modulated signals. And the Σ-Δ modulator (SDM) design — order, architecture, noise-shaping curve choice — differs per vendor, and that's the key variable determining listening differences.


3.1 DSD's Origins and Two Technical Routes


DSD was jointly proposed by Sony and Philips in 1996, but the two companies took completely different implementation routes.


The Sony route: Sony positioned DSD as a mastering-grade format from the start; SACD's DSD encoding directly used 1-bit/2.8224MHz Σ-Δ modulation. After 2005, Sony developed FPGA-based DSD remastering engines that can convert PCM to 5.6MHz (DSD128) or even 11.2MHz (DSD256) DSD in real time, with DSD digital filters removing ultrasonic noise in the digital domain.


The Philips route: Philips accumulated rich 1-bit conversion experience in its earlier Bitstream DAC technology. Its early "Bitstream" approach converted 16-bit signals into 1-bit streams with noise shaping and pulse-density modulation (PDM), becoming an important school of DAC design. In 2005, Philips' DSD division moved to Sonic Studio, whose Nexus/Nextage technology became one of the mainstream high-end mastering algorithms, favoring a relaxed, natural style.


3.1.1 A Third Force: Foobar2000 and foo_dsd_converter


Beyond Sony's and Philips' professional routes, there's a force from the open-source community and audiophiles — the Foobar2000 player and its foo_dsd_converter plugin.


foo_dsd_converter provides quite flexible real-time PCM→DSD conversion. It integrates multiple SDM modulator options (e.g., 5th/7th-order modulators, different noise-shaping curves) letting users tune upsampling parameters to taste. Combined with Foobar2000's rich plugin ecosystem, audiophiles can easily build a real-time PCM→DSD128/DSD256 chain — completely free.


Technically, foo_dsd_converter broke down the barrier to DSD conversion. SDM tuning that once required professional studios or high-end hardware vendors is now possible on any ordinary computer with Foobar2000. It has been instrumental in popularizing DSD playback — thousands of audiophiles first experienced the PCM-to-DSD listening difference through this plugin.


But there's a flip side: audiophile doesn't equal professional.


foo_dsd_converter's flexibility is both blessing and curse. With many parameters and no standardized guidance, many users pick mismatched settings: e.g., upsampling 44.1kHz CD rips to DSD256 with overly high modulator order, introducing modulation noise into the audible band; or wrongly configured noise-shaping curves, creating audible artifacts in the mid-high range.


The bigger problem is re-encoding and sharing. Open QQ-group cloud drives, Bilibili links, or forum shares: folders full of "DSD files" — labeled DSD64, DSD128, DSD256, often hundreds of GB. How many are native DSD recordings? How many are converted from PCM? What SDM parameters were used? What was the upsampling filter quality?


The answer: indistinguishable.


  • File labels don't record conversion origin or parameters. A folder marked "DSD128" could contain ten-year-old low-quality SDM conversions from MP3, or 24bit/192kHz masters converted with top algorithms — same filename, vastly different sound.
  • Many shared DSD files have parameter mismatches: sample rates incompatible with source (e.g., forcing 48kHz-family content through 44.1kHz-family DSD rates), producing fold-back distortion; wrong channel mapping; unreasonable gain settings.
  • Worse, some poor conversions leave audible high-frequency images at 20–30kHz due to improper noise-shaping parameters — possibly filtered in PCM playback but fully presented in DSD Direct playback, making first-time listeners think "DSD is bigger and more analog."

This "disorderly conversion + mass distribution" has objectively polarized DSD's reputation. Those who experienced quality SDM conversions praise DSD's purity and transparency; those who downloaded poor conversions may conclude "DSD is worse than CD." The divergence isn't about DSD technology itself — it's about source conversion quality and parameter discipline.


This is why more professional audio communities now urge: when sharing DSD files, note the conversion origin and parameters, even attach conversion logs. For quality-seeking audiophiles, rather than downloading DSD packs of unknown origin, convert in real time yourself from reliable PCM sources with quality SDM software (HQPlayer, DpdoEngine, etc.) — at least you know the quality of every step.


3.2 Core Modulator Parameter Differences


Though Σ-Δ modulators share the same mathematics, vendors make different choices on these parameters:


Order: A 7th-order Σ-Δ modulator is a common high-end configuration. Higher order means better in-band noise shaping — quantization noise pushed further, better audible-band SNR. But high-order modulators demand exponentially more from clock jitter and circuit precision: each additional order raises clock stability requirements by roughly 6dB. Excessively high order with an unstable clock source can introduce modulation noise or limit-cycle oscillation distortion. So higher isn't always better-sounding — system matching matters.


m-bit vs 1-bit: Beyond strict 1-bit DSD, Sony also defined DSD-Wide (m-bit SDM) — multi-bit (e.g., 6-bit or 8-bit) internal quantizers while preserving DSD-domain signal characteristics. DSD-Wide and 1-bit DSD coexist in the DSD domain and convert losslessly. Vendors handle m-bit very differently — some take the 8-bit path for higher in-band SNR, others 6-bit for faster modulation rates. These differences directly shape the final analog waveform details.


Noise Transfer Function (NTF): The NTF determines the frequency-domain distribution of quantization noise. Some vendors prefer a "gentle" curve pushing noise to ~50kHz; others choose "aggressive" curves to 100kHz+. The former sounds more natural and warm; the latter has higher in-band SNR but may cause intermodulation distortion as ultrasonic noise modulates back into the audible band. Fundamentally an engineering trade-off.


3.3 Representative Vendor SDM Routes


VendorSDM/DSD routeCore character
SonyFPGA-driven, DSD remaster engineUp to DSD256 (11.2MHz), refined and delicate, with DSD digital filter
Philips/Sonic StudioBitstream legacy → Nexus/NextageDirect lineage from original Bitstream DACs, relaxed and natural
dCSAll inputs upsampled to DSD128, proprietary RingDACPrecise and authoritative, full-FPGA, no conventional DAC chips
EMMLabsFPGA upsampling to 16× DSDFounder Ed Meitner, a key DSD advocate; lineage from the SACD era
Playback DesignsFPGA upsampling to 4× DSDTechnically close to EMMLabs but different tuning — emphasizes analog feel


Chapter 4: Upsampling — The "Re-creation" from PCM to DSD


When a DSD player or converter must handle non-native DSD PCM signals (CD rips, streaming PCM), it must first upsample the PCM, then feed it to the SDM for noise shaping. The filter design of this upsampling stage is another key variable determining the final sound.


4.1 Core Upsampling Filter Parameters


Tap count: The filter's tap count determines computational precision. More taps mean deeper stopband attenuation and more accurate signal reconstruction. High-end converter upsampling filters can reach hundreds of thousands to millions of taps. But more taps mean more latency and enormous FPGA/DSP compute demands.


Roll-off: An ideal low-pass filter stays flat in the passband and rolls off steeply before the stopband. But in reality, steeper rolloff means more time-domain ringing (pre/post-ringing). Some vendors favor steep rolloff for frequency-response precision (e.g., dCS); others choose gentle rolloff for more natural transients (many analog-styled designs).


Ring suppression: A filter's impulse response rings near the cutoff frequency — spurious oscillations before and after sharp signal changes. Excessive ringing makes sound "hard" and "digital," most evident on transient-rich sources like piano and percussion. Good upsampling algorithms balance frequency-domain precision against time-domain purity.


4.2 Vendor Upsampling Philosophies


Sony: The DSD remaster engine uses FPGAs with strong real-time compute. Its upsampling filters are specially optimized to suppress ringing while maintaining steep rolloff, paired with DSD digital filters that remove ultrasonic noise in the digital domain — the output is carefully shaped in both domains.


dCS: dCS is a recognized master of upsampling algorithms. Its latest platform DACs upsample all inputs (PCM or DSD) uniformly to DSD128, then convert via its proprietary 5-bit RingDAC. dCS upsampling uses proprietary filters with extremely high tap counts, plus five selectable mapping filters, letting users choose different upsampling curves by music genre or taste. dCS's sound character — precise imaging, open soundstage, authoritative bass — owes much to its superior upsampling.


EMMLabs & Playback Designs: Both descend from founder Ed Meitner's technical lineage, but have diverged. EMMLabs uses FPGAs to upsample everything to 16× DSD (45.1584MHz/49.152MHz) — the highest internal upsampling in consumer gear. Playback Designs stops at 4× DSD (11.2896MHz/12.288MHz), believing higher rates' theoretical advantages aren't audible and may introduce unnecessary processing noise. Both insist on staying in the DSD domain end to end, never converting DSD back to PCM.


Chord Electronics: Strictly speaking, Chord doesn't use DSD as an internal processing format; its FPGA-based pulse-array DACs (Hugo TT2, Dave) upsample all inputs to very high PCM rates (up to 2.048MHz) before pulse-array decoding. Founder Rob Watts argues DSD's 1-bit architecture is less optimal for DAC precision than high-bit-depth PCM upsampling — but Chord's algorithms also use noise shaping, fundamentally sharing DSD's noise-management approach, just via a different path.


Linn: Linn firmly backs PCM. All its DACs are pure-PCM, no native DSD decoding. Linn argues DSD's format efficiency is low and editing is difficult, while its own PCM upsampling (Masterpiece series) already achieves extremely low in-band noise and distortion. This route divergence itself shows: DSD isn't the only quality path — algorithm quality matters more than the format label.


4.3 How Upsampling and SDM Work Together


Upsampling and SDM don't operate independently — they're tightly coupled steps.


  • Upsampling determines the signal's time-frequency precision: tap count affects frequency resolution; rolloff affects time-domain purity.
  • SDM determines the noise's frequency distribution: order and NTF curve decide where quantization noise goes and how fast it decays.

Vendors' different trade-offs at each stage produce very different sound. A superb upsampling filter means little if the downstream SDM's noise shaping is weak — the in-band SNR advantage shrinks. Conversely, great SDM can't rescue severe ringing or phase distortion introduced during upsampling — the output still sounds "hard" or "digital."



Chapter 5: DAC Chip-Level Differences — ESS vs AKM's Different Philosophies


Even the same DSD file can sound very different through different DAC chips. This directly concerns how a DAC chip "digests" the DSD signal.


5.1 ESS (Sabre Series)


ESS DAC chips (ES9038PRO, ES9068AS, ES9039PRO, etc.) share a notable design trait: no dedicated independent 1-bit DSD path internally.


When fed a DSD signal, ESS chips upsample it through the internal DSP, convert it back to the PCM domain under the multi-bit HyperStream II architecture, and finally output via a multi-bit Δ-Σ modulator. In other words, when ESS plays DSD, the DSD signal is effectively "translated" into PCM inside the chip, then converted to analog.


This is controversial in both academia and audiophile circles. Supporters say ESS's internal algorithms are mature enough that the translation causes no audible information loss. Critics say DSD's core advantage — staying in the 1-bit pulse-density domain throughout — is broken by this process, subjecting the signal to unnecessary PCM-domain conversion that may add quantization error.


Note that ESS's newest architectures (e.g., ES9039PRO, 2022) improved the DSD path; some models now support DSD Direct mode, claiming DSD bypasses the internal DSP's multi-bit conversion straight into the modulator. But industry consensus is that these improvements are less "true DSD Direct" than optimization of the PCM-domain conversion down below the theoretical audibility threshold.


5.2 AKM (Velvet Sound Series)


AKM's design philosophy differs greatly from ESS. Its flagship chips (the AK4499EX/AK4191 split design, and earlier AK4497/AK4499) are more fully prepared for DSD processing.


In the AK4499EX/AK4191 architecture, AKM splits the traditional DAC into two parts:


  • AK4191 (digital front-end): digital signal reception, PCM upsampling, DSD signal conversion and modulation
  • AK4499EX (analog output): pure analog circuitry with a 128-element current-steering DAC array

This split physically isolates digital processing from analog output, greatly reducing digital noise crosstalk into the analog signal. For DSD, AKM's DAC array operates at 5.6/6.1MHz or 11.2/12.3MHz with excellent low-band SNR.


AKM offers DSD Direct mode, where the DSD signal skips most digital processing and goes straight to analog conversion by the physical DAC array — no PCM-domain intermediate translation. This is the implementation most DSD enthusiasts approve of: the 1-bit pulse stream stays in the DSD domain from source to analog output.


Additionally, AKM's newest architectures introduced asynchronous mode, where jitter depends only on the local crystal rather than the input signal's clock quality — especially beneficial with asynchronous transport protocols like USB.


5.3 Other Important DAC Approaches


TI/Burr-Brown (PCM17xx/DAC series): TI's DSD handling resembles AKM's, supporting DSD Direct mode (DSD Playback). Its newest chips (PCM1795, PCM5242, etc.) support native DSD decoding, though some lower-end models downsample high-rate DSD (above DSD256).


R-2R DACs (Holo Audio, Denafrips, etc.): R-2R resistor-ladder DACs are fundamentally multi-bit PCM architectures — natively incapable of 1-bit DSD. These vendors convert DSD via FPGA into multi-bit digital (usually 24-bit or 32-bit) before the R-2R resistor network converts to analog. Despite excellent conversion precision, strictly speaking DSD has been "translated" into PCM before entering the R-2R network.



Chapter 6: What This Means for Listening


After the technical analysis, back to a fundamental question: how do vendor differences in SDM, upsampling and DAC architecture ultimately affect what we hear?


6.1 Different Noise Distribution


Modulator order and noise-shaping curves determine where and how quantization noise is pushed into the ultrasonic range. Sony's FPGA engine pushes noise higher (DSD128/256 plus digital filters), theoretically optimal in-band SNR; the Philips-lineage Sonic Studio algorithms use gentler NTF curves, with noise decaying gradually across 30–50kHz, sounding more natural and relaxed.


At the chip level: ESS's "translating" DSD and AKM's "straight-through" DSD Direct produce completely different noise distributions. The former passes through a PCM-domain conversion, theoretically with slightly higher in-band noise (depending on internal algorithm quality); the latter stays in the 1-bit domain throughout, closer to the original DSD noise-shaping curve.


6.2 Different Transient Response


Upsampling filter tap count and rolloff determine impulse-response ringing, directly affecting "speed" and "aliveness":


  • High tap count, steep rolloff (e.g., dCS) → high frequency-domain precision but more time-domain ringing → detailed but possibly "hard," crisp instrumental outlines
  • Moderate taps, gentle rolloff (e.g., Playback Designs, Philips lineage) → higher time-domain purity, natural transients but slightly softer high extension → relaxed, non-fatiguing

DSD's 1-bit architecture inherently suffers less filter ringing than PCM, but ringing introduced at the upsampling stage still exists. This is why even in DSD playback, different devices' "speed" can differ hugely.


6.3 Different Tonal Directions


Combining SDM, upsampling and DAC implementation, brands form distinct tonal philosophies:


  • Sony's DSD remaster engine: "refined and delicate," clean and crisp background, open soundstage with clear imaging
  • Philips lineage (Sonic Studio/Nextage): "relaxed and natural," subtle warmth in the mids, soft high extension
  • dCS: "precise and authoritative," extremely accurate stable imaging, wide and deep but not over-rendered
  • EMMLabs: "transparent and direct," clean transparent background, dynamic and incisive, emphasizing information content
  • Playback Designs: "analog texture," dense but not overly sharp, vocals and strings with special fullness and sheen
  • Chord Electronics: though strictly not a DSD route, its "pulse array" character is extreme dynamic range and very low noise floor, neutral and transparent with moderate warmth
  • AKM-chip devices: thanks to DSD Direct, the 1-bit pulse-density signal is preserved most completely, cleaner noise distribution; mid-high air and overtone detail feel naturally rich
  • ESS-chip devices: because DSD is translated, mid-low density and impact often have distinctive character, though some users find high-frequency overtone air slightly behind AKM's DSD Direct

6.4 An Important Reminder


These listening differences aren't only between DSD and PCM — the same DSD file sounds more different across devices than different formats sound on the same device. This is crucial because it means:


  • If you heard DSD on an ESS device and the same PCM on an R-2R device, the difference you heard is mostly DAC architecture difference, not DSD-vs-PCM format difference.
  • Comparing DSD and PCM must be controlled on the same device, same DAC path — otherwise conclusions mislead.
  • High-end brands like dCS, EMMLabs and Playback Designs are revered in DSD playback not because DSD has "magic" but because they invest as much or more engineering into the DSD signal path as into PCM.


Chapter 7: A Complete Analogy — Seeing DSD's Value Through Image Processing


A visual analogy may help:


PCM is like a high-resolution digital photo, each pixel recording brightness and color in 16 or 24 bits. More pixels and deeper bit depth mean finer images. But even the best CMOS sensor has its own noise in shadow areas.


DSD's noise shaping is like applying a special denoising process to the same photo — not blurring the noise, but algorithmically pushing it into an ultra-high-frequency region the eye can't resolve, letting shadow detail emerge.


Different vendors' noise-shaping algorithms are like different image-denoising algorithms: some preserve texture but lose a trace of detail (warm-leaning), some are sharp and clear but may show slight artifacts (analytic-leaning), some are balanced but slow (complex-leaning).


DAC chip differences are like different displays — OLED, MicroLED, high-end IPS, each with its own color reproduction. The same denoised photo looks completely different on different monitors.


So when someone asks "is DSD better-sounding," the right answer isn't "yes" or "no" — it's "it depends on which SDM, which upsampling algorithm, and which DAC is playing it."



Chapter 8: Not Avoiding the Debate — DSD's Controversies and Limits


To be candid, DSD has real controversies and limits. For objectivity, let's face the following criticisms directly:


8.1 "DSD and PCM are indistinguishable in blind tests"


This is the most common objection, supported by several studies. The problem:

  • Most blind tests use DACs not specially optimized for DSD
  • Test samples usually cover only DSD64, not DSD128/256's high-frequency advantages
  • Listener variance is huge — some are extremely sensitive to noise-distribution changes, others entirely unaffected

Based on existing evidence, the reasonable conclusion: on ordinary playback systems, DSD vs PCM differences are indeed small — not all audiophiles can tell; on top-tier optimized systems, the difference is audible and stable — provided test conditions and listeners are screened.


8.2 "DSD files are big and hard to edit"


DSD64 is about 4× the size of CD-grade PCM (2.8M vs 1.4M rate difference), with DSD128/256 doubling again. Larger files demand more from storage and transport.


More critically, DSD's 1-bit nature makes digital-domain editing — volume, EQ, fades — difficult. Simple 1-bit-domain operations (like multiplying by a coefficient) introduce significant quantization distortion, so most DSD editing must happen in a multi-bit domain (8-bit DSD-Wide or PCM) and convert back. This adds mastering workload and quality-loss risk.


8.3 "True pure-DSD chains are extremely rare"


From recording to playback, a "pure DSD chain" is vanishingly rare in practice. Even workflows billed as "all-DSD" often enter multi-bit domains for mixing, mastering and gain adjustments. Only a few independent labels (Opus3, Blue Coast Records, etc.) truly maintain 1-bit DSD end to end from A/D to D/A.


Across the full chain, DAC-internal SDM quality, clock precision and power integrity often affect final sound more than whether the input file is PCM or DSD.


8.4 "DSD-to-PCM re-decoding is the common reality"


As noted, many DAC chips (especially ESS) and R-2R DACs convert DSD to PCM internally before decoding. On these systems, DSD input and PCM input actually share the same translation path — so the difference mainly comes from input DSD quality control (noise shaping done at recording/conversion) and internal translation quality.



Chapter 9: DSD's Future and Practical Recommendations


9.1 Technology Directions


  • Higher-rate DSD: DSD256 (11.2MHz) is standard in mid-high-end products; DSD512 (22.5792MHz/24.576MHz) appears in some flagships. Higher rates let noise-shaping curves be designed more "gently" — the ultrasonic band is wide enough that in-band quantization residue is lower.
  • Mature native DSD decoding: AKM's DSD Direct and Holo Audio's FPGA native-DSD paths make "pure DSD domain" decoding real.
  • Real-time conversion (ALL to DSD): pioneered by HQPlayer and others, real-time PCM-to-DSD is maturing. Users can convert any digital source to DSD128/256 live before the DAC. This usually sounds better than DAC-internal conversion because the real-time algorithms (HQPlayer's poly-sinc filters + SDM combinations) are specially optimized.

9.2 Practical Advice for Audiophiles


If you have an extremely high-quality recording (native DSD classical/jazz/vocals):


  • Prefer DACs with DSD Direct mode and FPGA native-DSD paths (AKM-chip devices, EMMLabs, Playback Designs, dCS)
  • Pair with quality real-time conversion tools (e.g., HQPlayer) to further exploit DSD noise shaping
  • Keep everything asynchronous (USB/network) to avoid clock jitter undermining SDM's theoretical advantages

If your source is mostly PCM (CD rips, streaming PCM):


  • Choose players/DACs with quality real-time conversion — examine the upsampling filter (tap count, rolloff, ring suppression)
  • When possible, try professional real-time PCM→DSD software like HQPlayer — its noise-shaping curves usually beat DAC-chip internal SDM
  • On a budget, you don't need DSD — a PCM DAC with excellent DSP algorithms (e.g., Chord) can reach very high playback quality

Don't fall into "format superstition":


  • A badly recorded DSD256 sounds worse than an excellently recorded CD rip (PCM 44.1kHz/16bit)
  • In blind tests, most people can't distinguish PCM 24/96 from DSD64 of the same recording — but on top-tier systems the difference is stably audible
  • Rather than spending heavily on DSD format, first ensure recording quality, clock precision and power integrity — those are the "Way"; format is the "technique."


Epilogue: DSD's Value Is Not in the Format, But in People


Back to the opening view: DSD is not magic. It can't fix bad recordings, can't compensate for system weaknesses, and may not even provide audible improvement on every system.


But this nearly 10,000-word technical breakdown wants to convey one core message: DSD's value isn't in the either/or question of "which is better, DSD or PCM" — it's that DSD reveals a profound truth in digital audio processing: how you manage quantization noise determines the quality ceiling of digital playback.


PCM is an honest but crude noise management: noise distributed uniformly, coexisting with the signal across the audible range. DSD's noise shaping is more refined management: via upsampling and modulation, pushing noise out of the sensitive hearing region, providing a cleaner playback environment for quality recordings.


And what truly enriches DSD playback is the different design decisions vendors made in SDM, upsampling and DAC implementation — Sony's FPGA engine, Philips's Bitstream legacy, dCS's RingDAC, AKM's DSD Direct, EMMLabs's all-DSD domain, Chord's pulse array… they are all different answers to the same "noise management" question.


So when you hear "DSD sounds better," the question worth asking isn't "is DSD really good?" — it's —


"Which vendor's SDM? Which upsampling algorithm? Which chip is decoding?"


The answer is often more meaningful than "DSD vs PCM" itself.



Lossless music + Hi-Res specs + DSD noise shaping — only these three combined form a complete experience upgrade. If your hard drive holds plenty of PCM lossless, try ALL-to-DSD real-time conversion and feel the calm transparency after quantization noise is "pushed away" — but remember, beneath that calm, the low-frequency rumble of traffic outside the studio window is still there. That's the true imprint of music's existence.


DSD is not a repairman; it's a purifier. It's not a magician; it's an engineer. And every choice — DSD or PCM, AKM or ESS, 7th-order SDM or R-2R — is a vendor's unique answer to the eternal question of "how to manage noise and restore music."


This is what makes DSD good — and the fundamental reason worth repeated listening and comparison.