Kenpath Labs

Input streaming

Send text over a WebSocket as it is produced and receive audio continuously.

When to use it

Use this API when the text is still being produced — typically an LLM’s token stream. Rather than buffering output into sentences and synthesizing one at a time, the server consumes fragments as they arrive and holds back only a few words: the model generates each audio chunk from everything already spoken plus a short lookahead, so it never needs a complete sentence.

  • Speech starts about ten words into the stream, mid-sentence: 0.86 s to first audio versus 1.92 s for sentence buffering on the same simulated LLM stream
  • Against a live LLM at typical API cadence, audio began ~450 ms after the model’s first token
  • Prosody is continuous across chunk seams — each chunk is generated with the preceding audio as context, so there are no restarts or intonation resets at chunk boundaries
  • The connection is opened once, not per utterance, so each reply skips DNS, TLS, and admission. Measured from an Indian client against production: first audio ~420 ms after connection open, token feed included

Two WebSocket endpoints expose this: a native one (recommended) and an ElevenLabs-compatible one for existing EL integrations. Both authenticate with your usual header on the upgrade request; see Authentication.

Native WebSocket

WS/v1/audio/speech/stream-input

Connect with query parameters, send text fragments as JSON messages the moment they’re produced, and read binary frames back: raw PCM, 16-bit LE mono at 24 kHz. The server chunks at safe boundaries and keeps rolling text-and-audio context, so chunk N+1 continues chunk N’s prosody. Don’t pre-chunk or sentence-split yourself.

ParameterTypeDefaultDescription
voicequery-Voice id from GET /v1/voices: an sv_-prefixed id, e.g. sv_enhdbrj5 (Aanya).
langquery-Language hint: any form works (hi, hin, hindi, hi-IN). Omit to auto-detect from script.
modequerysentenceChunking strategy: sentence (waits for sentence boundaries) or eager (word-level, lowest latency; see below).
speedquery1.0Speaking speed, 0.71.5, pitch-preserving. Applied continuously across the whole session, so chunk seams stay seamless.
temperature …query-Sampling knobs ride as query params too. Omit them; the server fills serving-tuned defaults.

Messages you send

ParameterTypeDefaultDescription
{"text": "fragment "}JSON-A text fragment, as small as a single token delta. Send them as fast as they arrive.
{"flush": true}JSON-Force everything buffered out as audio now (for example at the end of an LLM paragraph).
{"text": ""}JSON-End of stream. The server synthesizes the remainder, sends {"type": "done"}, and closes.
pip install git+https://github.com/kenpath-labs/svara-python.git
import asyncio
from openai import AsyncOpenAI
from svara import AsyncSvara
llm = AsyncOpenAI()
svara = AsyncSvara() # reads SVARA_API_KEY
async def speak(prompt: str):
async def deltas():
stream = await llm.chat.completions.create(
model="gpt-4o-mini", stream=True,
messages=[{"role": "user", "content": prompt}])
async for event in stream:
if event.choices[0].delta.content:
yield event.choices[0].delta.content
# One WebSocket; audio starts a few words into the LLM's output.
async for audio in svara.speech.stream_input(deltas(), voice="sv_enhdbrj5"):
player.feed(audio) # PCM16 @ 24 kHz mono
asyncio.run(speak("Explain how rainbows form, in three sentences."))

Eager mode

mode=eager is the lowest-latency configuration and maps directly onto how the model was trained. The server buffers incoming deltas at word granularity (mid-word token deltas are handled), and as soon as chunk_words + peek_words complete words exist it synthesizes the first chunk_words with the next few words attached as the lookahead peek:

ParameterTypeDefaultDescription
chunk_wordsquery4Words per synthesis chunk. Smaller = earlier first audio, more chunk seams (they’re seamless, but each chunk has scheduling overhead).
peek_wordsquery2Lookahead words held back so each chunk knows what’s coming (clamped 1–5). The final chunk goes out with no peek, which is the model’s end-of-utterance signal.
max_chunk_wordsquery20Upper bound on a chunk once text has queued up — keeps a fast LLM from producing one enormous chunk.

With defaults, speech starts roughly 6 words into the LLM’s output; typically well before the first sentence ends.

ElevenLabs-compatible WebSocket

WS/v1/text-to-speech/{voice_id}/stream-input

A drop-in implementation of the ElevenLabs realtime protocol, for existing EL integrations: the official EL SDKs’ WebSocket clients work unmodified. The full protocol is supported:

  • BOS: {"text": " ", "generation_config": {"chunk_length_schedule": [120,160,250,290]}} (schedule values clamped 50-500; auto_mode supported)
  • Text: {"text": "fragment ", "try_trigger_generation": true}, plus {"flush": true}; a bare space is a keepalive
  • EOS: {"text": ""}
  • Audio frames: {"audio": "<base64>", "alignment": …, "normalizedAlignment": …} in the negotiated output_format (query param, e.g. pcm_24000, mp3_44100_128), then {"isFinal": true} and a clean close (code 1000)
  • inactivity_timeoutquery param: 20 s default, 180 s max

Character timings in alignment are chunk-relative and approximate: chunk boundaries are sample-exact, characters uniform within a chunk.

Tips

  • Forward raw deltas. Don’t sentence-split, batch, or “clean up” the LLM stream client-side; the server’s chunker is the one that knows the model’s trained format.
  • Buffer ~150 ms on playback. A small jitter buffer on your audio sink absorbs network variance without hurting perceived latency.
  • Keep the socket warm for turns, not sessions. Open the connection when a reply starts, close on {"type": "done"}. Each connection is one utterance with one prosodic arc.
  • Concurrent conversations need concurrent streams: each open socket counts against your plan’s concurrency limit (see Rate limits).