Voice SDK for Speech & Telephony Audio

The AVBlocks Voice SDK is a cross-platform library for encoding, decoding and converting speech and voice audio. It is available for C++, .NET (C#, VB, F#) and Python, and runs on Windows, macOS and Linux.

It is designed for voice-recording, VoIP, telephony, call-logging, dictation and IVR applications that need compact, efficient speech codecs rather than full-fidelity music codecs.

Voice & speech codecs

Codec

Encode

Decode

Typical use

Opus

Modern VoIP, voice chat, WebRTC, low-latency speech

G.711 (A-Law / μ-Law)

Traditional telephony, PSTN, call recording

G.726 / ADPCM

Legacy telephony, voicemail, voice storage

PCM / LPCM / WAV

Raw / uncompressed voice capture and interchange

What you can build with the Voice SDK

  • Call recording / logging — capture and store telephony audio as G.711, G.726 or Opus.

  • VoIP and voice chat — encode low-latency speech with Opus.

  • IVR and voicemail systems — decode and play back G.711 / G.726 prompts and messages.

  • Dictation and transcription pipelines — record voice and convert to PCM for speech-to-text.

  • Format conversion — convert between Opus, G.711, G.726 and WAV.

Voice audio processing

  • Audio resampling (e.g. 8 kHz telephony to 16 kHz / 44.1 kHz)

  • Mono to stereo and stereo to mono

  • Change bitrate to control quality vs. size

Get started

Part of the Audio SDK. For music and general-purpose audio, see the Audio SDK; for video, see the Video SDK.