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import librosa | |
from transformers import Wav2Vec2ForCTC, AutoProcessor | |
import torch | |
import numpy as np | |
from huggingface_hub import hf_hub_download | |
from torchaudio.models.decoder import ctc_decoder | |
ASR_SAMPLING_RATE = 16_000 | |
ASR_LANGUAGES = {} | |
with open(f"data/asr/all_langs.tsv") as f: | |
for line in f: | |
iso, name = line.split(" ", 1) | |
ASR_LANGUAGES[iso.strip()] = name.strip() | |
MODEL_ID = "facebook/mms-1b-all" | |
processor = AutoProcessor.from_pretrained(MODEL_ID) | |
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID) | |
# lm_decoding_config = {} | |
# lm_decoding_configfile = hf_hub_download( | |
# repo_id="facebook/mms-cclms", | |
# filename="decoding_config.json", | |
# subfolder="mms-1b-all", | |
# ) | |
# with open(lm_decoding_configfile) as f: | |
# lm_decoding_config = json.loads(f.read()) | |
# # allow language model decoding for "eng" | |
# decoding_config = lm_decoding_config["eng"] | |
# lm_file = hf_hub_download( | |
# repo_id="facebook/mms-cclms", | |
# filename=decoding_config["lmfile"].rsplit("/", 1)[1], | |
# subfolder=decoding_config["lmfile"].rsplit("/", 1)[0], | |
# ) | |
# token_file = hf_hub_download( | |
# repo_id="facebook/mms-cclms", | |
# filename=decoding_config["tokensfile"].rsplit("/", 1)[1], | |
# subfolder=decoding_config["tokensfile"].rsplit("/", 1)[0], | |
# ) | |
# lexicon_file = None | |
# if decoding_config["lexiconfile"] is not None: | |
# lexicon_file = hf_hub_download( | |
# repo_id="facebook/mms-cclms", | |
# filename=decoding_config["lexiconfile"].rsplit("/", 1)[1], | |
# subfolder=decoding_config["lexiconfile"].rsplit("/", 1)[0], | |
# ) | |
# beam_search_decoder = ctc_decoder( | |
# lexicon=lexicon_file, | |
# tokens=token_file, | |
# lm=lm_file, | |
# nbest=1, | |
# beam_size=500, | |
# beam_size_token=50, | |
# lm_weight=float(decoding_config["lmweight"]), | |
# word_score=float(decoding_config["wordscore"]), | |
# sil_score=float(decoding_config["silweight"]), | |
# blank_token="<s>", | |
# ) | |
def transcribe(audio_data=None, lang="eng (English)"): | |
if not audio_data: | |
return "<<ERROR: Empty Audio Input>>" | |
if isinstance(audio_data, tuple): | |
# microphone | |
sr, audio_samples = audio_data | |
audio_samples = (audio_samples / 32768.0).astype(np.float32) | |
if sr != ASR_SAMPLING_RATE: | |
audio_samples = librosa.resample( | |
audio_samples, orig_sr=sr, target_sr=ASR_SAMPLING_RATE | |
) | |
else: | |
# file upload | |
isinstance(audio_data, str) | |
audio_samples = librosa.load(audio_data, sr=ASR_SAMPLING_RATE, mono=True)[0] | |
lang_code = lang.split()[0] | |
processor.tokenizer.set_target_lang(lang_code) | |
model.load_adapter(lang_code) | |
inputs = processor( | |
audio_samples, sampling_rate=ASR_SAMPLING_RATE, return_tensors="pt" | |
) | |
# set device | |
if torch.cuda.is_available(): | |
device = torch.device("cuda") | |
elif ( | |
hasattr(torch.backends, "mps") | |
and torch.backends.mps.is_available() | |
and torch.backends.mps.is_built() | |
): | |
device = torch.device("mps") | |
else: | |
device = torch.device("cpu") | |
model.to(device) | |
inputs = inputs.to(device) | |
with torch.no_grad(): | |
outputs = model(**inputs).logits | |
if lang_code != "eng" or True: | |
ids = torch.argmax(outputs, dim=-1)[0] | |
transcription = processor.decode(ids) | |
else: | |
assert False | |
# beam_search_result = beam_search_decoder(outputs.to("cpu")) | |
# transcription = " ".join(beam_search_result[0][0].words).strip() | |
return transcription | |
ASR_EXAMPLES = [ | |
["assets/english.mp3", "eng (English)"], | |
# ["assets/tamil.mp3", "tam (Tamil)"], | |
# ["assets/burmese.mp3", "mya (Burmese)"], | |
] | |
ASR_NOTE = """ | |
The above demo doesn't use beam-search decoding using a language model. | |
Checkout the instructions [here](https://huggingface.co/facebook/mms-1b-all) on how to run LM decoding for better accuracy. | |
""" |