livekit-plugins-gnani

LiveKit Agents plugin for Gnani — high-accuracy Speech-to-Text (Prisma) and low-latency Text-to-Speech (Timbre) for Indian languages.

Gnani.ai featuring Prisma (STT) and Timbre (TTS) models, supporting 10+ Indian languages with real-time streaming, multilingual transcription, and code-switching capabilities.

Installation

pip install livekit-plugins-gnani

This will also install the websockets and livekit-agents packages as dependencies.

Prerequisites

You need a Gnani API key. Email speechstack@gnani.ai to get started — all new accounts receive free credits, no credit card required.

Authentication

All APIs require a single API key — no organization_id or user_id needed.

Option 1 — Environment variable (recommended):

export GNANI_API_KEY="your-api-key"

Option 2 — Constructor argument:

stt = STT(api_key="your-api-key", language="hi-IN")
tts = TTS(api_key="your-api-key")

Migration note: If upgrading from an earlier version, remove any organization_id and user_id parameters — they are no longer accepted.

Quick Start

Speech-to-Text (REST + Streaming)

from livekit.plugins.gnani import STT

stt = STT(language="hi-IN")

# REST STT (file-based transcription)
speech_event = await stt.recognize(audio_buffer)

# Streaming STT (real-time WebSocket)
speech_stream = stt.stream()

Text-to-Speech

from livekit.plugins.gnani import TTS

# REST (default) - single-request batch synthesis
tts_rest = TTS(voice="Karan")

# SSE - chunked synthesis via Server-Sent Events (lower latency)
tts_sse = TTS(voice="Karan", synthesize_method="sse")

# WebSocket - chunked synthesis over WS (lowest latency)
tts_ws = TTS(voice="Karan", synthesize_method="websocket")

All three modes work with the standard LiveKit voice agent pipeline. The synthesize_method controls which transport synthesize() uses (REST, SSE, or WebSocket). The stream() method always uses WebSocket regardless of this setting.

Full Constructor Reference

STT — All parameters

from livekit.plugins.gnani import STT

stt = STT(
    language="en-IN",           # Default: "en-IN"
    sample_rate=16000,          # Default: 16000 (also: 8000)
    format="verbatim",          # Default: "verbatim" (also: "transcribe")
    preferred_language=None,    # Default: None
    itn_native_numerals=False,  # Default: False
    api_key=None,               # Default: None (reads GNANI_API_KEY env var)
    base_url="https://api.vachana.ai",  # Default
)

TTS — All parameters

from livekit.plugins.gnani import TTS

tts = TTS(
    voice="Karan",                    # Default: "Karan" (also: Simran, Nara, Riya, Viraj, Raju)
    model="vachana-voice-v3",         # Default: "vachana-voice-v3"
    sample_rate=16000,                # Default: 16000 (also: 8000, 22050, 44100)
    encoding="linear_pcm",           # Default: "linear_pcm" (also: "oggopus")
    container="wav",                  # Default: "wav" (also: "raw", "mp3", "mulaw", "ogg")
    num_channels=1,                   # Default: 1
    bitrate=None,                     # Default: None (also: "96k", "128k", "192k")
    synthesize_method="rest",         # Default: "rest" (also: "sse", "websocket")
    api_key=None,                     # Default: None (reads GNANI_API_KEY env var)
    base_url="https://api.vachana.ai",  # Default
)

Features

STT (Prisma)

• REST recognition — REST API (POST /stt/v3) for file-based transcription
• Real-time streaming — WebSocket API (wss://api.vachana.ai/stt/v3/stream) for live audio transcription with VAD
• 10+ Indian languages — see supported language codes
• Code-switching — supports multilingual and code-mixed audio
• Sample rates — 8 kHz and 16 kHz
• ITN support — Inverse Text Normalization via format="transcribe"

Streaming PCM Specification

All streaming audio must be sent as raw PCM binary frames — no container format (WAV, MP3) mid-stream.

Frames must be sent at real-time cadence. See STT Realtime — PCM Specification for full details.

TTS (Timbre)

• REST synthesis — single-request batch audio generation (synthesize_method="rest")
• SSE streaming — lower-latency chunked synthesis via Server-Sent Events (synthesize_method="sse")
• WebSocket synthesis — lowest-latency synthesis via synthesize_method="websocket" or the stream() method
• 6 voices — Karan, Simran, Nara, Riya, Viraj, Raju
• Configurable output — sample rate (8000–44100), encoding (linear_pcm, oggopus), container (raw, mp3, wav, mulaw, ogg)
• Runtime updates — change voice or model via update_options()

Supported Languages

STT Languages (Prisma)

Prisma uses BCP-47 locale codes (e.g. hi-IN). Supported:

• STT REST — Supported Languages
• STT Realtime — Supported Languages

TTS Languages (Timbre)

For the full list of supported languages, see TTS — Supported Languages.

Available Voices

Architecture

This plugin directly implements the Gnani REST and WebSocket APIs using aiohttp (for REST STT/TTS) and websockets (for streaming STT/TTS), adapting them into LiveKit's stt.STT and tts.TTS base classes. It uses the Prisma model for speech-to-text and the Timbre model for text-to-speech. No external SDK is required — all connection logic, authentication, and audio format handling is self-contained. Authentication uses a single api_key passed via the X-API-Key-ID header.

Documentation

• Gnani API Docs
• LiveKit Agents Docs
• Gnani STT Plugin Guide
• Gnani TTS Plugin Guide

License

Apache 2.0 — see LICENSE.