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Digital Audio

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Sound wave converted to bitstream — typically PCM (44.1/48/96 kHz). Film standard since DAT, now Dante over network. No tape saturation, but sample rate and bit depth matter.

Digital audio converts sound waves into bit sequences – this was the revolution that freed film sound from tape in the early 1990s. PCM (Pulse Code Modulation) is the standard method: continuous sound is measured at regular intervals (sampled) and stored as a numerical value. The sampling rate determines how often this measurement takes place per second. 44.1 kHz is sufficient for music, but on film sets, we work with 48 kHz – this has been the international standard since the DAT era. Some productions go up to 96 kHz, especially for high-resolution mastering or when submix headroom becomes critical.

The bit depth – usually 24 bits in professional environments – determines the resolution of these measured values. 24 bits theoretically mean approximately 144 dB dynamic range; 16 bits (CD standard) are significantly more limited but still sufficient for many applications. For sound recording on set, we almost exclusively use 24 bits / 48 kHz on recorders like the Sennheiser MKE 600 or wireless systems – this is the lowest common denominator for post-production.

The decisive advantage over analog tape: no degradation with copies, no wear, no tape distortion. You can copy the same WAV file a hundred times – it sounds identical. The problem lies not in the technology itself, but in quantization: when a value falls between two measurement points, it has to be rounded. This creates quantization noise. With 24 bits, it's inaudible; with 16 bits, it can become critical with very quiet signals.

In modern post-production, the Dante protocol dominates – digital audio over network, uncompressed, with minimal latency. This means an AES3 cable from set to cart is replaced by an Ethernet cable. Dante allows multi-channel streaming over a single network, saving time and cable clutter on larger productions. But this is also where the pitfalls begin: network stability becomes a critical factor. A dropout is a dropout – and you don't see it until you play back the takes.

Storage and archiving are another point: WAV is uncompressed, robust, Pro Tools compatible – the medium of choice. MP3 or AAC are for delivery, not for working copies. For color grading and sound design, lossless quality is non-negotiable. The workflow runs via session files (OMF, AAF) or simply via XML references to the media.

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