Update app.py

app.py

@@ -6,16 +6,18 @@ from speechbrain.pretrained import EncoderASR
 from transformers.file_utils import cached_path, hf_bucket_url
 
 cache_dir = './cache/'
-lm_file = hf_bucket_url("dragonSwing/wav2vec2-base-vn-270h", filename='4gram.zip')
+lm_file = hf_bucket_url(
+    "dragonSwing/wav2vec2-base-vn-270h", filename='4gram.zip')
 lm_file = cached_path(lm_file, cache_dir=cache_dir)
 with zipfile.ZipFile(lm_file, 'r') as zip_ref:
     zip_ref.extractall(cache_dir)
 lm_file = cache_dir + 'lm.binary'
 vocab_file = cache_dir + 'vocab-260000.txt'
-model = EncoderASR.from_hparams(source="dragonSwing/wav2vec2-base-vn-270h", 
+model = EncoderASR.from_hparams(source="dragonSwing/wav2vec2-base-vn-270h",
                                 savedir="/content/pretrained2/"
                                 )
 
+
 def get_decoder_ngram_model(tokenizer, ngram_lm_path, vocab_path=None):
     unigrams = None
     if vocab_path is not None:
@@ -37,19 +39,23 @@ def get_decoder_ngram_model(tokenizer, ngram_lm_path, vocab_path=None):
     decoder = build_ctcdecoder(vocab_list, ngram_lm_path, unigrams=unigrams)
     return decoder
 
+
 ngram_lm_model = get_decoder_ngram_model(model.tokenizer, lm_file, vocab_file)
 
+
 def transcribe_file(path, max_seconds=20):
     waveform = model.load_audio(path)
     if max_seconds > 0:
-      waveform = waveform[:max_seconds*16000]
+        waveform = waveform[:max_seconds*16000]
     batch = waveform.unsqueeze(0)
     rel_length = torch.tensor([1.0])
     with torch.no_grad():
         logits = model(batch, rel_length)
-    text_batch = [ngram_lm_model.decode(logit.detach().cpu().numpy(), beam_width=500) for logit in logits]
+    text_batch = [ngram_lm_model.decode(
+        logit.detach().cpu().numpy(), beam_width=500) for logit in logits]
     return text_batch[0]
 
+
 def speech_recognize(file_upload, file_mic):
     if file_upload is not None:
         file = file_upload
@@ -61,10 +67,18 @@ def speech_recognize(file_upload, file_mic):
     text = transcribe_file(file)
     return text
 
-inputs = [gr.inputs.Audio(source="upload", type='filepath', optional=True), gr.inputs.Audio(source="microphone", type='filepath', optional=True)]
-outputs =  gr.outputs.Textbox(label="Output Text")
+
+inputs = [gr.inputs.Audio(source="upload", type='filepath', optional=True), gr.inputs.Audio(
+    source="microphone", type='filepath', optional=True)]
+outputs = gr.outputs.Textbox(label="Output Text")
 title = "wav2vec2-base-vietnamese-270h"
 description = "Gradio demo for a wav2vec2 base vietnamese speech recognition. To use it, simply upload your audio, click one of the examples to load them, or record from your own microphone. Read more at the links below. Currently supports 16_000hz audio files"
 article = "<p style='text-align: center'><a href='https://huggingface.co/dragonSwing/wav2vec2-base-vn-270h' target='_blank'>Pretrained model</a></p>"
-examples=[['example1.wav', 'example1.wav'], ['example2.mp3', 'example2.mp3'], ['example3.mp3', 'example3.mp3'], ['example4.wav', 'example4.wav']]
-gr.Interface(speech_recognize, inputs, outputs, title=title, description=description, article=article, examples=examples).launch()
+examples = [
+    ['example1.wav', 'example1.wav'],
+    ['example2.mp3', 'example2.mp3'],
+    ['example3.mp3', 'example3.mp3'],
+    ['example4.wav', 'example4.wav'],
+]
+gr.Interface(speech_recognize, inputs, outputs, title=title,
+             description=description, article=article, examples=examples).launch()