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.