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import librosa
from transformers import Wav2Vec2ForCTC, AutoProcessor
import torch
import json
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] = name
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, lang="eng (English)"):
if isinstance(audio_data, tuple):
# microphone
sr, audio_samples = audio_data
audio_samples = (audio_samples/32768.0).astype(np.float)
print("case1", audio_samples[:5])
assert sr == ASR_SAMPLING_RATE, "Invalid sampling rate"
else:
# file upload
isinstance(audio_data, str)
print("case2 1", audio_data)
audio_samples = librosa.load(audio_fp, sr=ASR_SAMPLING_RATE, mono=True)[0]
print("case2", audio_samples[:5])
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.
"""
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