PSD 3100 HV

Most solid-state amplifiers operate their transistors across a wide portion of their transfer characteristic, which introduces non-linearity that must be corrected with large amounts of negative feedback — a technique that reduces distortion on paper but can compress dynamics and add its own sonic signature. T+A's approach is fundamentally different: by raising the supply voltage to 360V in power amplifiers (compared to the typical 60–80V), the music signal uses only the most linear 20% of each transistor's operating curve — the region where a transistor behaves almost identically to a vacuum tube. This means negative feedback can be reduced to near-zero levels, preserving the signal's original dynamics, micro-detail, and spatial information. Every HV circuit is built from discrete components rather than integrated circuits: individually selected and matched audio-grade J-FET transistors, hermetically sealed gold-contact relays for signal switching, and a precision discrete resistor ladder for volume control that maintains perfect channel balance and signal purity at every level.

PCM encodes amplitude values as multi-bit samples at fixed intervals, while DSD encodes audio as a continuous single-bit pulse density stream at extremely high sample rates. These are fundamentally different mathematical representations of sound, and converting one to the other before decoding inevitably discards information unique to each format. Most manufacturers use a single DAC chip that converts everything to one internal format — typically PCM. T+A refuses this compromise. Their Path Separation architecture provides two physically separate converter circuits sharing the same chassis: a Double-Differential Quadruple Converter optimised exclusively for PCM signals (supporting up to 32-bit/768 kHz), and a True 1-Bit converter that processes DSD natively without ever touching PCM. The incoming data stream is automatically routed to the correct path, ensuring that each format is decoded using the architecture best suited to its structure. This dual-path approach is significantly more expensive to implement, but it means both PCM and DSD listeners hear their music decoded in the optimal way.

DSD (Direct Stream Digital) was conceived as a way to capture audio with extreme temporal resolution — using a single bit toggling at rates from 2.8 MHz (DSD64) up to 49.2 MHz (DSD1024). The format's sonic character — its characteristic smoothness, analogue-like decay, and natural transient behaviour — comes directly from this pulse-density encoding. When a conventional DAC converts DSD to PCM for processing, it applies decimation filters that fundamentally alter the signal's temporal structure, losing exactly the qualities that make DSD distinctive. T+A's True 1-Bit converter avoids this entirely: the raw bitstream passes through a proprietary analogue reconstruction filter that converts the pulse density directly to an analogue waveform without any intermediate digital conversion. The result preserves the format's inherent musicality — the effortless fluidity and natural tonal warmth that DSD enthusiasts prize. With support up to DSD1024 in the Series 200 range, T+A handles the highest-resolution DSD recordings available today.

Every D/A converter chip, no matter how advanced, introduces small non-linearities and a noise floor inherent to its architecture. T+A's solution is statistical: by running four Burr-Brown 32-bit Sigma-Delta converters per channel in a double-differential configuration, the random errors of each individual chip are uncorrelated and tend to cancel each other out. The double-symmetrical topology — where pairs of converters operate in anti-phase — provides exceptional common-mode rejection, suppressing power supply noise, electromagnetic interference, and even-order harmonic distortion. The net result is a 6 dB reduction in the noise floor compared to a single converter — equivalent to gaining an extra bit of resolution — along with measurably lower distortion across the entire frequency range. With native support for PCM up to 32-bit/768 kHz, the converter handles the most demanding high-resolution formats while maintaining the ultra-low noise floor that reveals the finest musical details: the decay of a cymbal, the breath before a vocal phrase, the texture of a bowed string.

Digital audio reconstruction requires interpolating between discrete samples to recreate a continuous analogue waveform. The standard approach uses sinc-function FIR filters — mathematically perfect but prone to pre-ringing artefacts where the filter's impulse response precedes the actual transient, smearing the leading edge of percussive sounds. T+A developed their Bezier interpolation as a fundamentally different approach: instead of sinc functions, it uses Bezier polynomial curves — the same mathematics used in computer graphics to draw smooth curves through control points — to interpolate between samples. The Pure Bezier setting produces zero pre-ringing and zero post-ringing, preserving the exact temporal structure of transients for what many listeners describe as the most natural, analogue-like sound. The four selectable filters let the listener choose their priority: FIR Long for textbook-flat frequency response, FIR Short for a balance of frequency accuracy and improved timing, Bezier/IIR for analogue warmth with minimal ringing, or Pure Bezier for absolute temporal accuracy. This level of user control over the reconstruction algorithm is virtually unique in the industry.

In any device that combines digital processing with analogue audio — streamers, DACs, integrated amplifiers — the high-speed digital circuits generate electromagnetic noise that can couple into the sensitive analogue signal path through shared ground planes, power supply rails, and even physical proximity. Most manufacturers attempt to manage this with careful PCB layout and filtering, but some coupling is inevitable when digital and analogue circuits share a common electrical ground. T+A's DASS eliminates this problem at the most fundamental level: ultra-fast Silicon Labs digital isolators transfer the audio data across a galvanic barrier — there is literally no electrical connection between the digital and analogue sections. Each domain has its own dedicated mains transformer and completely independent power supply, meaning digital switching noise has no conductive path to reach the analogue circuitry. The digital isolators are fast enough to pass high-resolution audio data without introducing jitter or timing errors. The audible benefit is a lower noise floor and a blacker background, revealing low-level detail and spatial cues that would otherwise be masked by digital contamination.