python/README.md
A fast, accurate, on-device AI library for building interactive voice applications. Join our Discord to get help and support.
pip install moonshine-voice
# Listens to the microphone, logging to the console when there are
# speech updates.
moonshine-voice mic
Installing the package adds a moonshine-voice command (with a shorter
moonshine alias) that groups the built-in tools as subcommands: mic,
transcribe, tts, agent, download, and g2p. Run
moonshine-voice --help, or moonshine-voice <command> --help for a specific
tool. Each subcommand is equivalent to python -m moonshine_voice.<module>, so
either invocation style works.
"""Transcribes live audio from the default microphone"""
import time
from moonshine_voice import MicTranscriber
# MicTranscriber handles connecting to the microphone, capturing the audio
# data, detecting voice activity, breaking the speech up into segments,
# transcribing it, and calling you back as the results firm up over time.
mic = (
MicTranscriber()
.on_text(lambda text: print(f"\r{text}", end="", flush=True))
.on_line(lambda line: print(f"\r{line.text}"))
)
# Downloads the model files and caches them, so the first call is the slow one.
mic.load()
mic.start()
print("Listening to the microphone, press Ctrl+C to stop...")
while True:
time.sleep(0.1)
on_text gives you the line as it is being spoken, rewritten as the model
changes its mind, and on_line gives you each line once it is final.
Configuration is chainable and everything has a working default:
language(), model_arch(), device(), update_interval(), and
on_progress() for a download bar.
When you need more than the text, such as line ids, speaker spans, or word timings, attach a listener object instead. Both styles can be used together.
from moonshine_voice import MicTranscriber, TranscriptEventListener
class TestListener(TranscriptEventListener):
def on_line_started(self, event):
print(f"Line started: {event.line.text}")
def on_line_text_changed(self, event):
print(f"Line text changed: {event.line.text}")
def on_line_completed(self, event):
print(f"Line completed: {event.line.text}")
mic = MicTranscriber()
mic.add_listener(TestListener())
mic.load()
mic.start()
If you have a different source you're capturing audio from you can supply it directly to a transcriber.
"""Transcribes live audio from an arbitrary audio source."""
from moonshine_voice import (
Transcriber,
TranscriptEventListener,
get_model_for_language,
load_wav_file,
get_assets_path,
)
import os
from typing import Iterator, Tuple
def audio_chunk_generator(
wav_file_path: str, chunk_duration: float = 0.1
) -> Iterator[Tuple[list, int]]:
"""
Example function that loads a WAV file and yields audio chunks.
This demonstrates how you can integrate your own proprietary
audio data capture sources. Replace this function with your own
implementation that yields (audio_chunk, sample_rate) tuples.
Args:
wav_file_path: Path to the WAV file to load
chunk_duration: Duration of each chunk in seconds
Yields:
Tuple of (audio_chunk, sample_rate) where:
- audio_chunk: List of float audio samples
- sample_rate: Sample rate in Hz
"""
audio_data, sample_rate = load_wav_file(wav_file_path)
chunk_size = int(chunk_duration * sample_rate)
for i in range(0, len(audio_data), chunk_size):
chunk = audio_data[i: i + chunk_size]
yield (chunk, sample_rate)
model_path, model_arch = get_model_for_language("en")
transcriber = Transcriber(
model_path=model_path, model_arch=model_arch)
stream = transcriber.create_stream(update_interval=0.5)
stream.start()
class TestListener(TranscriptEventListener):
def on_line_started(self, event):
print(f"{event.line.start_time:.2f}s: Line started: {event.line.text}")
def on_line_text_changed(self, event):
print(
f"{event.line.start_time:.2f}s: Line text changed: {event.line.text}")
def on_line_completed(self, event):
print(f"{event.line.start_time:.2f}s: Line completed: {event.line.text}")
listener = TestListener()
stream.add_listener(listener)
# Feed audio chunks from the generator into the stream.
wav_file_path = os.path.join(get_assets_path(), "two_cities.wav")
for chunk, sample_rate in audio_chunk_generator(wav_file_path):
stream.add_audio(chunk, sample_rate)
stream.stop()
stream.close()
Voice commands go through AgentFlow. Register the phrases you want to listen
for and it handles the rest: it downloads the speech recognition, speech
synthesis and phrase-matching models, opens the microphone, matches what the
user said semantically rather than by exact wording, and runs your handler.
from moonshine_voice import AgentFlow
def lights_on(d):
print("\nš” LIGHTS ON!")
def lights_off(d):
print("\nš LIGHTS OFF!")
runner = (
AgentFlow()
.always("turn on the lights", lights_on)
.always("turn off the lights", lights_off)
)
load() downloads and opens everything it needs, and start_listening() opens
the microphone. There's nothing else to construct:
runner.load()
runner.start_listening()
try:
while True:
time.sleep(0.1)
except KeyboardInterrupt:
print("\n\nStopping...", file=sys.stderr)
finally:
runner.close()
Configuration is chainable, and everything has a working default ā language(),
voice(), trigger_threshold(), on_heard() / on_said() / on_error() for
observing the conversation, and use_mic_transcriber() / use_text_to_speech()
when you'd rather supply your own. To drive a runner from text instead of audio,
turn the microphone off with microphone(False) and feed it handle_utterance().
For multi-turn conversations ā asking a question, confirming an answer,
spelling out a password ā register a flow with listen_for() instead of a
global. See examples/python/agent_flow.py, or run moonshine-voice agent.
TextToSpeech follows the same shape: configure it, load() it, then use it.
from moonshine_voice import TextToSpeech
tts = TextToSpeech().language("en_us").voice("kokoro_af_heart")
tts.load()
tts.say("Hello from Moonshine.")
tts.wait()
say() returns immediately and queues the utterance, pre-synthesizing the next
one while the current one plays, so consecutive calls run together without a
gap. wait() blocks until the queue drains, stop() cancels it, and
is_talking() polls. If you would rather have the samples than hear them, use
synthesize(). The chainable setters are language(), voice(),
output_device(), volume(), models_from(), and on_progress() for a
download bar.
To speak in someone else's voice, clone_from() takes a few seconds of them
talking, either as a .wav file or as a (pcm, sample_rate) pair.
tts = TextToSpeech().language("en_us").cloning()
tts.load()
tts.clone_from("some-speech.wav")
tts.say("Now I sound like you.")
cloning() fetches ZipVoice and clone ASR during load() so clone_from()
only swaps the reference clip. Call it before load(); without it,
clone_from() / start_cloning() raise. To capture the reference clip from
the microphone instead, start_cloning() hands back a VoiceClone that
listens until it has heard enough usable speech.
clone = tts.start_cloning()
clone.on_ready(lambda: print("Got it, you can stop talking."))
clone.from_microphone()
tts.clone_from(clone)
The framework currently supports English, Spanish, Mandarin, Japanese, Korean, Vietnamese, Arabic, and Ukrainian. We are working on wider language support, and you can see which are supported in your version by calling supported_languages(). To use a language, request it using get_model_for_language() passing in the two-letter language code. For example get_model_for_language("es") will download the Spanish models and pass the information you need to create Transcriber objects using them.
For more information, see the main Moonshine Voice documentation.
The code and English-language models are released under the MIT License - see the main project repository for details. The models used for other languages are released under the Moonshine Community License.