SLN to OPUS Converter

SLN (Signed Linear) is a headerless raw audio format storing 16-bit signed linear PCM samples at 8000 Hz mono, most closely associated with Asterisk — the open-source PBX framework developed by Digium (now Sangoma Technologies). Within Asterisk, SLN serves as the native internal audio representation: every codec transcoding operation passes through signed linear as an intermediate step. This makes SLN the backbone of Asterisk's codec translation architecture. The format contains nothing but raw samples — no headers, no metadata, no framing — so parameters must be known in advance. While this lack of self-description might seem limiting, it is actually an advantage in telephony where sample format is fixed by convention and every overhead byte matters across thousands of simultaneous channels. The 8000 Hz rate aligns with the G.711 standard for traditional telephony, capturing the full 300-3400 Hz voice band. Asterisk also supports extended variants (sln16, sln32, sln48) for wideband audio. SLN files require no decoding — just direct memory mapping — making them ideal for real-time mixing, conferencing, and prompt playback in high-density VoIP environments.

Opus is a versatile, open audio codec standardized by the IETF as RFC 6716 in 2012. It fuses two coding approaches — SILK for speech and CELT for music — into one algorithm that blends between them based on content type and bitrate. This hybrid design lets Opus outperform virtually every other codec across a wide range of uses: low-latency voice at 6 kbps, high-fidelity music at 128 kbps, and everything in between. It supports bitrates from 6 to 510 kbps, sample rates up to 48 kHz, and frame sizes as small as 2.5 ms, giving it the lowest algorithmic latency of any mainstream audio codec. Three advantages make Opus especially compelling. It is completely royalty-free and open-source, removing licensing barriers that hold back proprietary codecs. It achieves transparent quality at roughly half the bitrate of MP3 and beats AAC at equivalent rates. And its low latency makes it the mandatory codec for WebRTC, so every modern browser ships with an Opus decoder. WhatsApp, Discord, Zoom, and YouTube all rely on Opus for real-time audio.