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Easy-to-use, low-latency speech-to-text library for realtime applications
RealtimeSTT listens to the microphone and transcribes voice into text.
It's ideal for:
Hint: Check out RealtimeTTS, the output counterpart of this library, for text-to-voice capabilities. Together, they form a powerful realtime audio wrapper around large language models.
This library uses:
These components represent the "industry standard" for cutting-edge applications, providing the most modern and effective foundation for building high-end solutions.
pip install RealtimeSTT
This will install all the necessary dependencies, including a CPU support only version of PyTorch.
Additional steps are needed for a GPU-optimized installation. These steps are recommended for those who require better performance and have a compatible NVIDIA GPU.
To use RealtimeSTT with GPU support via CUDA please follow these steps:
Install NVIDIA CUDA Toolkit 11.8:
Install NVIDIA cuDNN 8.7.0 for CUDA 11.x:
Install PyTorch with CUDA support:
pip uninstall torch
pip install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu118
Note: To check if your NVIDIA GPU supports CUDA, visit the official CUDA GPUs list.
Basic usage:
Start and stop of recording are manually triggered.
recorder.start()
recorder.stop()
print(recorder.text())
Recording based on voice activity detection.
recorder = AudioToTextRecorder()
print(recorder.text())
Keyword activation before detecting voice. Write the comma-separated list of your desired activation keywords into the wake_words parameter. You can choose wake words from these list: alexa, americano, blueberry, bumblebee, computer, grapefruits, grasshopper, hey google, hey siri, jarvis, ok google, picovoice, porcupine, terminator.
recorder = AudioToTextRecorder(wake_words="jarvis")
print('Say "Jarvis" then speak.')
print(recorder.text())
You can set callback functions to be executed on different events (see Configuration) :
def my_start_callback():
print("Recording started!")
def my_stop_callback():
print("Recording stopped!")
recorder = AudioToTextRecorder(on_recording_started=my_start_callback,
on_recording_finished=my_stop_callback)
The test subdirectory contains a set of scripts to help you evaluate and understand the capabilities of the RealtimeTTS library.
simple_test.py
wakeword_test.py
translator.py
pip install openai realtimetts
.openai_voice_interface.py
pip install openai realtimetts
.advanced_talk.py
pip install openai keyboard realtimetts
.minimalistic_talkbot.py
pip install openai realtimetts
.AudioToTextRecorder
When you initialize the AudioToTextRecorder
class, you have various options to customize its behavior.
model (str, default="tiny"): Model size or path for transcription.
language (str, default=""): Language code for transcription. If left empty, the model will try to auto-detect the language.
on_recording_start: A callable function triggered when recording starts.
on_recording_stop: A callable function triggered when recording ends.
on_transcription_start: A callable function triggered when transcription starts.
spinner (bool, default=True): Provides a spinner animation text with information about the current recorder state.
level (int, default=logging.WARNING): Logging level.
silero_sensitivity (float, default=0.6): Sensitivity for Silero's voice activity detection ranging from 0 (least sensitive) to 1 (most sensitive). Default is 0.6.
webrtc_sensitivity (int, default=3): Sensitivity for the WebRTC Voice Activity Detection engine ranging from 1 (least sensitive) to 3 (most sensitive). Default is 3.
post_speech_silence_duration (float, default=0.2): Duration in seconds of silence that must follow speech before the recording is considered to be completed. This ensures that any brief pauses during speech don't prematurely end the recording.
min_gap_between_recordings (float, default=1.0): Specifies the minimum time interval in seconds that should exist between the end of one recording session and the beginning of another to prevent rapid consecutive recordings.
min_length_of_recording (float, default=1.0): Specifies the minimum duration in seconds that a recording session should last to ensure meaningful audio capture, preventing excessively short or fragmented recordings.
pre_recording_buffer_duration (float, default=0.2): The time span, in seconds, during which audio is buffered prior to formal recording. This helps counterbalancing the latency inherent in speech activity detection, ensuring no initial audio is missed.
wake_words (str, default=""): Wake words for initiating the recording. Multiple wake words can be provided as a comma-separated string. Supported wake words are: alexa, americano, blueberry, bumblebee, computer, grapefruits, grasshopper, hey google, hey siri, jarvis, ok google, picovoice, porcupine, terminator
wake_words_sensitivity (float, default=0.6): Sensitivity level for wake word detection (0 for least sensitive, 1 for most sensitive).
wake_word_activation_delay (float, default=0): Duration in seconds after the start of monitoring before the system switches to wake word activation if no voice is initially detected. If set to zero, the system uses wake word activation immediately.
wake_word_timeout (float, default=5): Duration in seconds after a wake word is recognized. If no subsequent voice activity is detected within this window, the system transitions back to an inactive state, awaiting the next wake word or voice activation.
on_wakeword_detected: A callable function triggered when a wake word is detected.
on_wakeword_timeout: Callback function to be called when the system goes back to an inactive state after when no speech was detected after wake word activation.
on_wakeword_detection_start: Callback function to be called the system starts to listen for wake words
on_wakeword_detection_end: Callback function to be called when stopping to listen for wake words (e.g. because of timeout or wake word detected)
Contributions are always welcome!
MIT
Kolja Beigel
Email: kolja.beigel@web.de
GitHub