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Delete asr.py(auto/audio)
Browse files- asr.py(auto/audio) +0 -59
asr.py(auto/audio)
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import librosa
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import torch
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import numpy as np
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import langid # Language detection library
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from transformers import Wav2Vec2ForCTC, AutoProcessor
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ASR_SAMPLING_RATE = 16_000
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MODEL_ID = "facebook/mms-1b-all"
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# Load MMS Model
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processor = AutoProcessor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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model.eval()
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def detect_language(text):
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"""Detects language using langid (fast & lightweight)."""
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lang, _ = langid.classify(text)
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return lang if lang in ["en", "sw"] else "en" # Default to English
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def transcribe_audio(audio_data=None):
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if not audio_data:
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return "<<ERROR: Empty Audio Input>>"
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# Process Microphone Input
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if isinstance(audio_data, tuple):
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sr, audio_samples = audio_data
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audio_samples = (audio_samples / 32768.0).astype(np.float32)
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if sr != ASR_SAMPLING_RATE:
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audio_samples = librosa.resample(audio_samples, orig_sr=sr, target_sr=ASR_SAMPLING_RATE)
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# Process File Upload Input
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else:
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if not isinstance(audio_data, str):
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return "<<ERROR: Invalid Audio Input>>"
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audio_samples = librosa.load(audio_data, sr=ASR_SAMPLING_RATE, mono=True)[0]
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inputs = processor(audio_samples, sampling_rate=ASR_SAMPLING_RATE, return_tensors="pt")
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# **Step 1: Transcribe without Language Detection**
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with torch.no_grad():
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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raw_transcription = processor.decode(ids)
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# **Step 2: Detect Language from Transcription**
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detected_lang = detect_language(raw_transcription)
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lang_code = "eng" if detected_lang == "en" else "swh"
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# **Step 3: Reload Model with Correct Adapter**
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processor.tokenizer.set_target_lang(lang_code)
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model.load_adapter(lang_code)
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# **Step 4: Transcribe Again with Correct Adapter**
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with torch.no_grad():
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outputs = model(**inputs).logits
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ids = torch.argmax(outputs, dim=-1)[0]
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final_transcription = processor.decode(ids)
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return f"Detected Language: {detected_lang.upper()}\n\nTranscription:\n{final_transcription}"
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