kotoba-sdk 0.1.1

Python SDK for Kotoba Speech APIs (REST transcription jobs + WebSocket streaming ASR / TTS / S2ST)

kotoba-sdk

Python SDK for Kotoba speech APIs — REST batch transcription and streaming ASR, TTS, and speech-to-speech translation over WebSockets.

Phase-1 alpha. See docs/quickstart.md.

Install

pip install kotoba-sdk

Or from a checkout:

git clone https://github.com/kotoba-tech/kotoba-python.git
cd kotoba-python
uv venv
uv pip install -e .

Python ≥ 3.10. Optional mic extra (pip install 'kotoba-sdk[mic]') installs sounddevice for live-microphone examples.

Configure endpoints

The SDK reads configuration from these env vars only — set the ones for the routes you actually need:

Variable	Purpose
KOTOBA_API_KEY	Bearer token sent as Authorization: Bearer … (REST + WS)
KOTOBA_ASR_REST_URL	REST API base URL including version prefix, e.g. https://.../v1
KOTOBA_ASR_URL	WebSocket URL for live ASR, e.g. wss://.../asr
KOTOBA_TTS_JA_URL	WebSocket URL for Japanese TTS, e.g. wss://.../tts
KOTOBA_S2ST_EN_JA_URL	WebSocket URL for English-to-Japanese speech translation

You can also register routes from code:

import kotoba
kotoba.register_endpoint("tts", None, "ko", "wss://.../tts")

URLs passed explicitly via url=... on a call take precedence over the registry.

Quickstart

import kotoba

client = kotoba.KotobaClient()  # reads KOTOBA_API_KEY + KOTOBA_*_URL from env

# 1) Speech recognition (REST batch — default for files)
result = client.asr.transcribe(
    "examples/audio/ja/example.mp3", language="ja"
)
print(result.text)

# 2) Text-to-Speech (Japanese, default speaker)
audio = client.tts.synthesize("こんにちは、世界。", language="ja")
audio.to_wav("hello.wav")

# 3) Speech-to-Speech translation (English -> Japanese)
translated = client.s2st.translate(
    "examples/audio/en/example.mp3", src="en", tgt="ja"
)
translated.to_wav("translated.wav")
print(translated.transcript_source)

KotobaClient() reads its credentials and URLs from env vars. Pass them explicitly if you'd rather not rely on the environment:

client = kotoba.KotobaClient(
    api_key="sk_...",
    url="https://.../v1",                  # REST base
    asr_ws_url="wss://.../asr",
    tts_ja_ws_url="wss://.../tts",
    s2st_en_ja_ws_url="wss://.../sts",
)

Streaming (the live surface)

ASR, TTS, and S2ST are all streaming-first. Audio chunks and partial transcripts surface the moment the server emits them, so you can play / display incrementally instead of waiting for the full response.

Streaming output

ASR streams transcript deltas as audio arrives; TTS streams audio chunks as the server produces them from a single text prompt. ASR accepts a generator of PCM16 chunks on the input side (feed + drain run concurrently); TTS sends the full text in one frame and streams audio back:

# ASR: pcm16 bytes in -> transcript deltas out
for delta in client.asr.transcribe_stream(mic_chunks(), language="ja"):
    print(delta, end="", flush=True)

# TTS: full text in -> pcm audio chunks streamed out
for pcm in client.tts.synthesize_stream("こんにちは、世界。", language="ja"):
    speaker.write(pcm)

Async (recommended for production)

import asyncio, kotoba

async def main():
    client = kotoba.AsyncKotobaClient()

    async with client.tts.stream(language="ja") as session:
        await session.synthesize("こんにちは。本日はよろしくお願いします。")

        async for event in session:
            if event.type == "audio_chunk":
                await play(event.audio)
            elif event.type == "done":
                break

asyncio.run(main())

Sync (notebooks, scripts)

import kotoba

client = kotoba.KotobaClient()
with client.s2st.stream(src="en", tgt="ja") as session:
    for chunk in pcm16_chunks_from_mic():
        session.send_audio(chunk)
    session.commit()
    for event in session:
        if event.type == "partial_transcript":
            print(event.text, end="", flush=True)
        elif event.type == "audio_chunk":
            speaker.write(event.audio)
        elif event.type == "done":
            break

The sync wrapper runs an asyncio loop on a background daemon thread, so the underlying transport is identical — only the call style differs.

What's in the box

Module	What
kotoba.KotobaClient / AsyncKotobaClient	Top-level entry point
client.asr.transcribe(path, ...)	REST batch transcription with optional with_timestamps=True
client.asr.stream(...) / transcribe_stream(iter)	Streaming ASR (Japanese, English) over WebSocket
client.tts.stream(...) / synthesize(...) / synthesize_stream(...)	Streaming TTS (Japanese)
client.s2st.stream(...) / translate(...)	Streaming speech-to-speech translation
kotoba.register_endpoint(...)	Add (modality, src, tgt) -> URL routes
kotoba.audio.*	PCM16 / float32 WAV helpers

Examples

Each example under examples/ is runnable with uv run examples/<file>.py and uses bundled audio under examples/audio/ by default.

File	What it shows	Required env
asr_rest_sync.py	REST batch transcription with with_timestamps=True, sync	KOTOBA_API_KEY, KOTOBA_ASR_REST_URL
asr_rest_async.py	Same, async with AsyncKotobaClient context manager	KOTOBA_API_KEY, KOTOBA_ASR_REST_URL
asr_stream_async.py	Live ASR via transcribe_stream(generator) with first-token-latency measurement	KOTOBA_API_KEY, KOTOBA_ASR_URL
tts_synthesize_sync.py	One-shot TTS with explicit speaker_id	KOTOBA_API_KEY, KOTOBA_TTS_JA_URL
tts_stream_async.py	One-shot text in → streamed audio chunks with first-audio-latency timing	KOTOBA_API_KEY, KOTOBA_TTS_JA_URL
s2st_stream_async.py	File in → live transcript + translated WAV out	KOTOBA_API_KEY, KOTOBA_S2ST_EN_JA_URL
s2st_mic_async.py	Live microphone in → translated WAV out (Ctrl-C to stop). Requires pip install 'kotoba-sdk[mic]' and PortAudio.	KOTOBA_API_KEY, KOTOBA_S2ST_EN_JA_URL

REST is shown in both sync + async because the context-manager pattern matters for resource cleanup. Streaming examples are async-by-default — wrap with kotoba.KotobaClient() for sync (the snippets above show the conversion).

Public API

kotoba.KotobaClient / kotoba.AsyncKotobaClient

KotobaClient(
    *,
    api_key: str | None = None,           # KOTOBA_API_KEY
    url: str | None = None,               # KOTOBA_ASR_REST_URL  (REST)
    asr_ws_url: str | None = None,        # KOTOBA_ASR_URL       (WS ASR)
    tts_ja_ws_url: str | None = None,     # KOTOBA_TTS_JA_URL    (WS TTS)
    s2st_en_ja_ws_url: str | None = None, # KOTOBA_S2ST_EN_JA_URL
    timeout: float = 30.0,                # per-request HTTP timeout (s)
    max_retries: int = 3,                 # for 429/5xx and network errors
)

Exposes:

• .asr — ASRClient / AsyncASRClient (REST + WS)
• .tts — TTSClient / AsyncTTSClient (WS)
• .s2st — S2STClient / AsyncS2STClient (WS)

The async variant supports async with … and exposes await client.close().

client.asr.transcribe(...) — REST batch helper

transcribe(
    audio_file_path: str | Path,
    *,
    language: str = "ja",
    with_timestamps: bool = False,  # ask server for per-segment timestamps
    poll_interval: float = 1.0,     # initial GET polling interval (s)
    poll_backoff: float = 1.5,      # multiplied each poll
    max_poll_interval: float = 10.0,
    timeout: float = 1200.0,        # overall deadline for job completion
) -> TranscriptResult

POSTs the file, polls GET /transcription_jobs/{id} with exponential backoff, returns the final transcript. Raises TranscriptionError on server-reported failure, TimeoutError if the deadline elapses.

When with_timestamps=True, TranscriptResult.segments is populated with [Segment(text, start, end), …].

Low-level REST helpers

client.asr.submit_job(path, language="ja") -> JobIDResponse  # POST
client.asr.get_job(job_id)                -> JobStatus       # GET, 202→processing

JobStatus.state is one of JobState.processing | done | error. For done, read .transcription; for error, read .error_message.

WebSocket entry points

client.asr.stream(language="ja", url=...)           -> ASRSession
client.asr.transcribe_stream(audio_iter, ...)       -> Iterator[str]

client.tts.stream(language="ja", speaker_id=..., url=...)  -> TTSSession
client.tts.synthesize_stream(text, ...)                    -> Iterator[bytes]
client.tts.synthesize(text, ...)                           -> AudioResult

client.s2st.stream(src="en", tgt="ja", url=...)  -> S2STSession
client.s2st.translate(path, src="en", tgt="ja")  -> S2STResult

URLs resolve from the per-route env vars (KOTOBA_ASR_URL, KOTOBA_TTS_JA_URL, KOTOBA_S2ST_EN_JA_URL) unless passed explicitly with url=.

Exceptions

All inherit from kotoba.KotobaError:

Exception	When
AuthError	HTTP 401/403, WS auth rejection
ProtocolError	Other 4xx, or a server error frame violating the contract
APIError	Transport or 5xx that exhausted retries
TimeoutError	HTTP timeout, WS handshake timeout, or transcribe() polling deadline exceeded
JobNotFoundError	GET returned 404
TranscriptionError	Job completed in error state
UnsupportedRouteError	No WS URL registered for the requested (modality, src, tgt)

Retry behavior (REST)

Both sync and async clients retry on network errors, 429, and 5xx with exponential backoff. Retry-After headers on 429 are honored (async client). 4xx other than 429 raise immediately.

Development

uv venv
uv pip install -e ".[dev]"
uv run pytest