generic-audio-transcriber

Drop-in speech-to-text for any Python project. Audio bytes in, JSON-ready text out.

from audio_transcriber import transcribe

result = transcribe(audio_bytes)   # mp3, wav, webm, ogg, m4a... detected automatically
print(result.text)                 # "Hello, this is a test."
print(result.to_json())            # {"text": "...", "language": "en", ...}

No API keys, no servers, no cloud. It runs locally on the CPU, and the model is downloaded automatically the first time you use it.

Why

Many apps need the same small feature: the user sends or records audio, and the app needs the text. This package is that feature as one function call, so you can add it to a project without learning anything about speech recognition.

Under the hood it uses faster-whisper (OpenAI's Whisper model), which is accurate, supports about 99 languages, and runs well on a plain CPU. You never have to train or tune a model.

Install

pip install git+https://github.com/EduardoMilani8/generic-audio-transcriber.git

Requires Python 3.9+. You do not need to install ffmpeg: audio decoding is bundled.

Usage

From bytes (the main use case)

from audio_transcriber import transcribe

result = transcribe(audio_bytes)

From a file path or file object

transcribe("recording.mp3")

with open("recording.wav", "rb") as f:
    transcribe(f)

Inside a web endpoint

from fastapi import FastAPI, UploadFile
from audio_transcriber import transcribe

app = FastAPI()

@app.post("/transcribe")
async def endpoint(file: UploadFile):
    return transcribe(await file.read()).to_dict()

The result

result.text                  # full transcript
result.language              # detected language code, e.g. "pt"
result.language_probability  # confidence of the language detection
result.duration              # audio length in seconds
result.segments              # timed pieces: .start, .end, .text
result.to_dict()             # plain dict
result.to_json()             # JSON string (non-ASCII kept readable)

Silent audio gives an empty text, not an error.

Options

transcribe(
    audio,
    model="small",         # tiny | base | small | medium | large-v3
    language=None,         # "pt", "en", ... None = auto-detect
    device="cpu",          # "cpu" | "cuda" | "auto"
    compute_type="int8",   # "int8" is light; "float16" suits GPUs
    beam_size=5,
    vad_filter=True,       # skip silence, reduces made-up text
)

Choosing a model is a trade-off between weight and accuracy:

The model is loaded once per process and reused, so only the first call is slow. Setting language explicitly is faster and more accurate than auto-detection.

Errors

from audio_transcriber import transcribe, TranscriptionError

try:
    transcribe(data)
except TranscriptionError as e:
    ...  # empty input or audio that cannot be decoded

Command line

audio-transcriber recording.mp3            # prints JSON
audio-transcriber recording.mp3 --text     # prints plain text
audio-transcriber recording.mp3 --model base --language pt
cat recording.mp3 | audio-transcriber -    # read bytes from stdin

Development

python -m venv .venv && source .venv/bin/activate
pip install -e ".[dev]"
pytest                                # fast unit tests, no model needed
RUN_SLOW=1 pytest -m slow             # end-to-end tests, downloads the tiny model

Notes

• The default device is cpu because it works on every machine. Pass device="cuda" if you have a working CUDA setup and want GPU speed.
• av is pinned below version 19 because faster-whisper still uses an argument that newer releases removed. The pin can be lifted once faster-whisper fixes it.

License

MIT