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Create asr.py
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asr.py
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import librosa
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from transformers import Wav2Vec2ForCTC, AutoProcessor
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import torch
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import numpy as np
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from pathlib import Path
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from huggingface_hub import hf_hub_download
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from torchaudio.models.decoder import ctc_decoder
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ASR_SAMPLING_RATE = 16_000
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ASR_LANGUAGES = {}
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with open(f"data/asr/all_langs.tsv") as f:
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for line in f:
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iso, name = line.split(" ", 1)
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ASR_LANGUAGES[iso.strip()] = name.strip()
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MODEL_ID = "facebook/mms-1b-all"
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processor = AutoProcessor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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# lm_decoding_config = {}
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# lm_decoding_configfile = hf_hub_download(
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# repo_id="facebook/mms-cclms",
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# filename="decoding_config.json",
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# subfolder="mms-1b-all",
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# )
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# with open(lm_decoding_configfile) as f:
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# lm_decoding_config = json.loads(f.read())
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# # allow language model decoding for "eng"
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# decoding_config = lm_decoding_config["eng"]
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# lm_file = hf_hub_download(
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# repo_id="facebook/mms-cclms",
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# filename=decoding_config["lmfile"].rsplit("/", 1)[1],
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# subfolder=decoding_config["lmfile"].rsplit("/", 1)[0],
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# )
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# token_file = hf_hub_download(
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# repo_id="facebook/mms-cclms",
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# filename=decoding_config["tokensfile"].rsplit("/", 1)[1],
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# subfolder=decoding_config["tokensfile"].rsplit("/", 1)[0],
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# )
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# lexicon_file = None
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# if decoding_config["lexiconfile"] is not None:
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# lexicon_file = hf_hub_download(
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# repo_id="facebook/mms-cclms",
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# filename=decoding_config["lexiconfile"].rsplit("/", 1)[1],
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# subfolder=decoding_config["lexiconfile"].rsplit("/", 1)[0],
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# )
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# beam_search_decoder = ctc_decoder(
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# lexicon=lexicon_file,
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# tokens=token_file,
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# lm=lm_file,
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# nbest=1,
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# beam_size=500,
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# beam_size_token=50,
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# lm_weight=float(decoding_config["lmweight"]),
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# word_score=float(decoding_config["wordscore"]),
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# sil_score=float(decoding_config["silweight"]),
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# blank_token="<s>",
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# )
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def transcribe(audio_data=None, lang="eng (English)"):
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if not audio_data:
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return "<<ERROR: Empty Audio Input>>"
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if isinstance(audio_data, tuple):
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# microphone
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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(
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audio_samples, orig_sr=sr, target_sr=ASR_SAMPLING_RATE
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)
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else:
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# file upload
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if not isinstance(audio_data, str):
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return "<<ERROR: Invalid Audio Input Instance: {}>>".format(type(audio_data))
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audio_samples = librosa.load(audio_data, sr=ASR_SAMPLING_RATE, mono=True)[0]
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lang_code = lang.split()[0]
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processor.tokenizer.set_target_lang(lang_code)
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model.load_adapter(lang_code)
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inputs = processor(
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audio_samples, sampling_rate=ASR_SAMPLING_RATE, return_tensors="pt"
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)
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# set device
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if torch.cuda.is_available():
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device = torch.device("cuda")
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elif (
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hasattr(torch.backends, "mps")
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and torch.backends.mps.is_available()
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and torch.backends.mps.is_built()
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):
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device = torch.device("mps")
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else:
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device = torch.device("cpu")
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model.to(device)
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inputs = inputs.to(device)
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with torch.no_grad():
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outputs = model(**inputs).logits
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if lang_code != "eng" or True:
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ids = torch.argmax(outputs, dim=-1)[0]
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transcription = processor.decode(ids)
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else:
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assert False
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# beam_search_result = beam_search_decoder(outputs.to("cpu"))
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# transcription = " ".join(beam_search_result[0][0].words).strip()
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return transcription
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ASR_EXAMPLES = [
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["upload/english.mp3", "eng (English)"],
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# ["upload/tamil.mp3", "tam (Tamil)"],
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# ["upload/burmese.mp3", "mya (Burmese)"],
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]
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ASR_NOTE = """
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The above demo doesn't use beam-search decoding using a language model.
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Checkout the instructions [here](https://huggingface.co/facebook/mms-1b-all) on how to run LM decoding for better accuracy.
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"""
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