PyThaiASR

Python Thai Automatic Speech Recognition

PyThaiASR is a Python package for Automatic Speech Recognition with focus on Thai language. It have offline thai automatic speech recognition model.

License: Apache-2.0 License

Google Colab: Link Google colab

Install

pip install pythaiasr

By default, PyThaiASR uses Typhoon ASR powered by ONNX Runtime for low-latency, lightweight offline and realtime speech recognition on CPU and GPU. On first use, Typhoon model files are automatically downloaded to ~/pythaiasr-data/typhoon-asr-realtime/.

For PyTorch & Transformers models (Wav2Vec2 / Whisper): If you want to use the Wav2Vec2 or Whisper models:

pip install pythaiasr[torch]

For Wav2Vec2 with language model: If you want to use wannaphong/wav2vec2-large-xlsr-53-th-cv8-* with a language model:

pip install pythaiasr[lm]
pip install https://github.com/kpu/kenlm/archive/refs/heads/master.zip

For live audio streaming: If you want to stream live audio from your microphone:

pip install pythaiasr[stream]

For Sherpa-ONNX Diarization Backend (Optional): PyThaiASR includes a built-in ONNX diarization engine out-of-the-box requiring no extra dependencies. If you prefer using the optional Sherpa-ONNX backend:

pip install pythaiasr[diarize]

Usage

File-based ASR

from pythaiasr import asr

file = "sample.wav"

# Uses Typhoon ASR (FastConformer RNN-T ONNX) by default
print(asr(file))

# With timestamps (returns dictionary with 'text', 'chunks', and 'timestamps')
result = asr(file, return_timestamps=True)
print(result["text"])
for chunk in result["chunks"]:
    print(f"[{chunk['start']:.2f}s -> {chunk['end']:.2f}s] {chunk['text']}")

# Or explicitly select another model (requires pythaiasr[torch])
# print(asr(file, model="airesearch/wav2vec2-large-xlsr-53-th"))
# print(asr(file, model="biodatlab/whisper-small-th-combined"))
# print(asr(file, model="biodatlab/whisper-th-medium-timestamp", return_timestamps=True))

Live Audio Streaming

Stream audio directly from your microphone/soundcard in real-time:

from pythaiasr import stream_asr

# Streams audio in real-time using Typhoon ASR by default
for transcription in stream_asr():
    print(transcription, end=" ", flush=True)
    # Press Ctrl+C to stop

And examples/stream_example.py

Real-Time Streaming from File or Microphone

from pythaiasr import FastConformerRNNT, RealtimeStreamASR, stream_from_file, stream_from_mic

model = FastConformerRNNT(device="auto")
streamer = RealtimeStreamASR(model=model, step_sec=0.48)

# Simulate streaming from a pre-recorded audio file
stream_from_file(streamer, "sample.wav")

# Or stream live from microphone with sounddevice
# stream_from_mic(streamer)

Speech Diarization (Who Spoke When)

Detect speaker turns and timestamps using ONNX:

from pythaiasr import diarize

# Identify who spoke when
segments = diarize("meeting.wav")
for seg in segments:
    print(f"[{seg['start']:.2f}s - {seg['end']:.2f}s] {seg['speaker']}")

Speech Diarization + ASR (asr_diarize)

Detect speakers and transcribe each speaker turn with ASR:

from pythaiasr import asr_diarize

# Attributed transcription per speaker
turns = asr_diarize("meeting.wav", asr_model="typhoon_asr")
for turn in turns:
    print(f"[{turn['start']:.2f}s - {turn['end']:.2f}s] {turn['speaker']}: {turn['text']}")

See examples/diarize_example.py

============================================================
1. Speech Diarization (Who Spoke When)
============================================================
Processing: examples/../tests/test-diarize.wav ...
[  0.17s ->   1.87s] SPEAKER_02
[  1.97s ->   4.30s] SPEAKER_01
[  4.83s ->   6.49s] SPEAKER_02
[  6.78s ->   8.36s] SPEAKER_01

============================================================
2. Diarization + Speech Recognition (ASR Diarize)
============================================================
Transcribing turns with Typhoon ASR: examples/../tests/test-diarize.wav ...
[  0.17s ->   1.87s] SPEAKER_02: สวัสดีชาวโลกทุกท่าน
[  1.97s ->   4.30s] SPEAKER_01: แล้วระบบนี้ทํางานอย่างไร
[  4.83s ->   6.49s] SPEAKER_02: ใช้ปัญญาประดิษฐ์ในการทดสอบ
[  6.78s ->   8.36s] SPEAKER_01: ใช้งานได้ดีทีเดียว

API

asr

asr(
    data: Union[str, np.ndarray],
    model: str = _model_name,
    lm: bool = False,
    device: str = None,
    sampling_rate: int = 16_000,
    return_timestamps: Optional[Union[bool, str]] = None,
    timestamps: Optional[Union[bool, str]] = None,
)

• data: path of sound file or numpy array of the voice
• model: The ASR model (default: typhoon_asr)
• lm: Use language model (for wav2vec2 models with LM)
• device: device (auto, cpu, cuda)
• sampling_rate: The sample rate
• return_timestamps: Return timestamps dictionary (True, "word", or "char")
• timestamps: Alias for return_timestamps
• return: Thai text from ASR (str) or dictionary with "text", "chunks", and "timestamps" if return_timestamps=True

stream_asr

stream_asr(
    model: str = _model_name,
    lm: bool = False,
    device: str = None,
    chunk_duration: float = None,
    sampling_rate: int = 16_000,
    return_timestamps: bool = False,
    timestamps: Optional[bool] = None,
)

• model: The ASR model (default: typhoon_asr)
• lm: Use language model (for wav2vec2 models with LM)
• device: device for running model
• chunk_duration: Duration of each audio chunk in seconds (default: 0.48s for Typhoon, 5.0s for others)
• sampling_rate: The sample rate (default: 16000)
• return_timestamps: Yield dictionary with text and chunk timestamps (True or False)
• timestamps: Alias for return_timestamps
• yield: Thai text transcription (or dict with timestamp) from each audio chunk

Options for model

• typhoon_asr / typhoon-asr-realtime (default) - Typhoon FastConformer RNN-T ONNX model (offline & realtime)
• airesearch/wav2vec2-large-xlsr-53-th - AI RESEARCH - PyThaiNLP model (requires pythaiasr[torch])
• wannaphong/wav2vec2-large-xlsr-53-th-cv8-newmm - Thai Wav2Vec2 with CommonVoice V8 (newmm tokenizer) (requires pythaiasr[torch])
• wannaphong/wav2vec2-large-xlsr-53-th-cv8-deepcut - Thai Wav2Vec2 with CommonVoice V8 (deepcut tokenizer) (requires pythaiasr[torch])
• biodatlab/whisper-small-th-combined - Thai Whisper small model (requires pythaiasr[torch])
• biodatlab/whisper-th-medium-combined - Thai Whisper medium model (requires pythaiasr[torch])
• biodatlab/whisper-th-large-combined - Thai Whisper large model (requires pythaiasr[torch])
• biodatlab/whisper-th-medium-timestamp - Thai Whisper medium model with timestamp support (requires pythaiasr[torch])

You can read about models from the list:

• typhoon-ai/typhoon-asr-realtime / wannaphong/typhoon-asr-realtime-onnx - Typhoon FastConformer RNN-T ONNX model
• airesearch/wav2vec2-large-xlsr-53-th - AI RESEARCH - PyThaiNLP model
• annaphong/wav2vec2-large-xlsr-53-th-cv8-newmm - Thai Wav2Vec2 with CommonVoice V8 (newmm tokenizer) + language model
• wannaphong/wav2vec2-large-xlsr-53-th-cv8-deepcut - Thai Wav2Vec2 with CommonVoice V8 (deepcut tokenizer) + language model
• biodatlab/whisper-small-th-combined - Thai Whisper small model
• biodatlab/whisper-th-medium-combined - Thai Whisper medium model
• biodatlab/whisper-th-large-combined - Thai Whisper large model
• biodatlab/whisper-th-medium-timestamp - Thai Whisper medium model with timestamp support

diarize

diarize(
    data: Union[str, Path, np.ndarray],
    model: str = "pyannote_segmentation",
    device: Optional[str] = None,
    sampling_rate: int = 16_000,
    num_speakers: Optional[int] = None,
    min_speakers: Optional[int] = None,
    max_speakers: Optional[int] = None,
    onset: float = 0.5,
    offset: float = 0.5,
    min_duration_on: float = 0.3,
    min_duration_off: float = 0.5,
    backend: str = "onnx",
) -> List[Dict[str, Union[float, str]]]

• data: Audio file path or 1D numpy array of audio waveform.
• model: Diarization model identifier (default: "pyannote_segmentation").
• device: Device to run inference on ("auto", "cpu", "cuda").
• sampling_rate: Audio sampling rate (default: 16000).
• num_speakers: Exact number of speakers if known.
• onset: Speech onset probability threshold (default: 0.5).
• offset: Speech offset probability threshold (default: 0.5).
• min_duration_on: Minimum speaker turn duration in seconds (default: 0.3).
• min_duration_off: Minimum silence duration to split turns in seconds (default: 0.5).
• backend: Diarization engine ("onnx" or "sherpa-onnx", default: "onnx").
• Returns: List of segments with start, end, and speaker keys.

asr_diarize

asr_diarize(
    data: Union[str, Path, np.ndarray],
    asr_model: str = "typhoon_asr",
    diarize_model: str = "pyannote_segmentation",
    device: Optional[str] = None,
    sampling_rate: int = 16_000,
    lm: bool = False,
    num_speakers: Optional[int] = None,
    merge_same_speaker: bool = True,
    max_merge_gap: float = 0.5,
    backend: str = "onnx",
    **kwargs,
) -> List[Dict[str, Union[float, str]]]

• data: Audio file path or 1D numpy array of audio waveform.
• asr_model: The ASR model name (default: "typhoon_asr").
• diarize_model: Diarization model name (default: "pyannote_segmentation").
• merge_same_speaker: Whether to merge adjacent speech turns from the same speaker (default: True).
• max_merge_gap: Maximum gap in seconds between same-speaker segments to merge (default: 0.5).
• backend: Diarization engine ("onnx" or "sherpa-onnx", default: "onnx").
• Returns: List of speaker turns with start, end, speaker, and text keys.

Docker

To use this inside of Docker do the following:

docker build -t <Your Tag name> .
docker run docker run --entrypoint /bin/bash -it <Your Tag name>

You will then get access to a interactive shell environment where you can use python with all packages installed.