raphaelmerx commited on
Commit
d690b2a
·
1 Parent(s): 37d60b5

Transcribe in chunks

Browse files

Avoid OOM on large audio files

Files changed (1) hide show
  1. app.py +28 -16
app.py CHANGED
@@ -1,6 +1,7 @@
1
  import gradio as gr
2
  from transformers import Wav2Vec2ForCTC, AutoProcessor
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  import torch
 
4
  import librosa
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  import json
6
 
@@ -24,20 +25,30 @@ def transcribe(audio_file_mic=None, audio_file_upload=None, language="English (e
24
  # Make sure audio is 16kHz
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  speech, sample_rate = librosa.load(audio_file)
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  if sample_rate != 16000:
 
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  speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000)
28
 
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- # Keep the same model in memory and simply switch out the language adapters by calling load_adapter() for the model and set_target_lang() for the tokenizer
 
 
 
 
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  language_code = iso_codes[language]
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  processor.tokenizer.set_target_lang(language_code)
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  model.load_adapter(language_code)
33
 
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- inputs = processor(speech, sampling_rate=16_000, return_tensors="pt")
 
 
 
 
 
35
 
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- with torch.no_grad():
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- outputs = model(**inputs).logits
 
38
 
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- ids = torch.argmax(outputs, dim=-1)[0]
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- transcription = processor.decode(ids)
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  return transcription
42
 
43
  examples = [
@@ -50,14 +61,15 @@ examples = [
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  description = '''Automatic Speech Recognition with [MMS](https://ai.facebook.com/blog/multilingual-model-speech-recognition/) (Massively Multilingual Speech) by Meta.
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  Supports [1162 languages](https://dl.fbaipublicfiles.com/mms/misc/language_coverage_mms.html). Read the paper for more details: [Scaling Speech Technology to 1,000+ Languages](https://arxiv.org/abs/2305.13516).'''
52
 
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- iface = gr.Interface(fn=transcribe,
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- inputs=[
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- gr.Audio(source="microphone", type="filepath", label="Record Audio"),
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- gr.Audio(source="upload", type="filepath", label="Upload Audio"),
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- gr.Dropdown(choices=languages, label="Language", value="English (eng)")
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- ],
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- outputs=gr.Textbox(label="Transcription"),
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- examples=examples,
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- description=description
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- )
 
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  iface.launch()
 
1
  import gradio as gr
2
  from transformers import Wav2Vec2ForCTC, AutoProcessor
3
  import torch
4
+ import numpy as np
5
  import librosa
6
  import json
7
 
 
25
  # Make sure audio is 16kHz
26
  speech, sample_rate = librosa.load(audio_file)
27
  if sample_rate != 16000:
28
+ print('resampling')
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  speech = librosa.resample(speech, orig_sr=sample_rate, target_sr=16000)
30
 
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+ # Cut speech into chunks
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+ chunk_size = 30 * 16000 # 30s * 16000Hz
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+ chunks = np.split(speech, np.arange(chunk_size, len(speech), chunk_size))
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+
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+ # load model adapter for this language
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  language_code = iso_codes[language]
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  processor.tokenizer.set_target_lang(language_code)
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  model.load_adapter(language_code)
39
 
40
+ transcriptions = []
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+ for chunk in chunks:
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+ inputs = processor(chunk, sampling_rate=16_000, return_tensors="pt")
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+
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+ with torch.no_grad():
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+ outputs = model(**inputs).logits
46
 
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+ ids = torch.argmax(outputs, dim=-1)[0]
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+ transcription = processor.decode(ids)
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+ transcriptions.append(transcription)
50
 
51
+ transcription = ' '.join(transcriptions)
 
52
  return transcription
53
 
54
  examples = [
 
61
  description = '''Automatic Speech Recognition with [MMS](https://ai.facebook.com/blog/multilingual-model-speech-recognition/) (Massively Multilingual Speech) by Meta.
62
  Supports [1162 languages](https://dl.fbaipublicfiles.com/mms/misc/language_coverage_mms.html). Read the paper for more details: [Scaling Speech Technology to 1,000+ Languages](https://arxiv.org/abs/2305.13516).'''
63
 
64
+ iface = gr.Interface(
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+ fn=transcribe,
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+ inputs=[
67
+ gr.Audio(source="microphone", type="filepath", label="Record Audio"),
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+ gr.Audio(source="upload", type="filepath", label="Upload Audio"),
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+ gr.Dropdown(choices=languages, label="Language", value="English (eng)")
70
+ ],
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+ outputs=gr.Textbox(label="Transcription"),
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+ examples=examples,
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+ description=description
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+ )
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  iface.launch()