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Digital audio is defined by two numbers: how often the signal is sampled (sample rate) and how finely each sample is measured (bit depth). Everything else — file format, lossless versus lossy, "hi-res" badges — is built on top of those two. Understanding what each controls separates meaningful resolution from marketing.

Sample Rate

Key Spec

Sample Rate (kHz)

The sample rate sets the highest frequency the signal can represent. By the Nyquist theorem, the captured bandwidth is half the sample rate: CD’s 44.1 kHz captures up to 22.05 kHz, comfortably beyond the ~20 kHz limit of human hearing. Higher rates (96, 192, 384 kHz) extend bandwidth into the ultrasonic — useful as production headroom and for gentler anti-alias filtering, but not audibly necessary for playback, since the extra content is above what anyone can hear.

Bit Depth

Key Spec

Bit Depth (bits)

Bit depth sets the dynamic range — the span between the quietest and loudest representable signal — at roughly 6.02 dB per bit. 16-bit yields about 96 dB of dynamic range (CD quality); 24-bit yields about 144 dB, far exceeding any listening room’s noise floor or any DAC’s actual performance. The practical value of 24-bit is production headroom — room to set levels and dither without audible noise — not a wider range you can actually hear at playback.

The math, simply: Dynamic range ≈ 6.02 × bit depth. Captured bandwidth = sample rate ÷ 2. 16-bit/44.1 kHz already exceeds the limits of human hearing in both dimensions. Higher numbers help during recording and mastering; their playback benefit is mostly theoretical.

Lossless vs Lossy

Format Class

Lossless (FLAC, ALAC, WAV, AIFF)

Lossless formats reconstruct the original PCM data bit-for-bit. FLAC and ALAC compress the file (typically 40–60%) without discarding any audio data — they unpack to the exact original. WAV and AIFF are uncompressed PCM. For archival and critical listening, lossless is the standard: it guarantees you hear exactly what was on the master.

Format Class

Lossy (MP3, AAC, Ogg Vorbis)

Lossy formats permanently discard data the encoder judges inaudible, achieving much smaller files. At high bitrates (256–320 kbps MP3, 256 kbps AAC), the loss is transparent to most listeners on most material. At low bitrates, artefacts (smearing, pre-echo, dulled high frequencies) become audible. Lossy is ideal for portable and bandwidth-limited use; it is not appropriate for archiving or repeated re-encoding.

PCM vs DSD

Encoding

PCM vs DSD

PCM (pulse-code modulation) measures amplitude at each sample — the dominant format for production and playback. DSD (Direct Stream Digital), used on SACD, encodes audio as a 1-bit stream at very high rates (2.8 MHz for DSD64 and up). DSD can sound excellent but is harder to edit and process, so most DSD is converted to PCM somewhere in the chain anyway. Neither is inherently superior at playback; mastering quality matters far more than the encoding.

"High-Resolution" and MQA

"High-resolution audio" generally means anything exceeding CD’s 16-bit/44.1 kHz — commonly 24-bit/96 kHz or 24-bit/192 kHz. The badge guarantees the container, not the source: a hi-res file is only as good as the master it was made from, and an upsampled CD master gains nothing. MQA is a proprietary format that folds high-res information into a smaller file using a licensed decoder; it remains controversial because it is lossy and requires compatible hardware. The single biggest determinant of how a recording sounds is the quality of the master, not its resolution number.

Quick Reference: Bit Depth and Sample Rate

FormatDynamic RangeBandwidthTypical Use
16-bit / 44.1 kHz~96 dBto 22.05 kHzCD, the transparency baseline
24-bit / 96 kHz~144 dBto 48 kHzHi-res streaming and downloads
24-bit / 192 kHz~144 dBto 96 kHzStudio masters, archival
DSD64~120 dB (audio band)to ~50 kHz usableSACD, audiophile downloads
320 kbps MP3Source-limitedto ~20 kHzTransparent portable use

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