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Update app.py
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app.py
CHANGED
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@@ -28,7 +28,7 @@ device = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
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print(f"Bruker enhet: {device}")
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@spaces.GPU(duration=60 * 2)
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def pipe(file, return_timestamps=False):
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asr = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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@@ -52,7 +52,8 @@ def format_output(text):
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text = re.sub(r'(\.{3,}|[.!:?])', lambda m: m.group() + '<br>', text)
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return text
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def transcribe(file, return_timestamps=False):
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waveform, sample_rate = torchaudio.load(file)
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audio_duration = waveform.size(1) / sample_rate
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@@ -66,19 +67,34 @@ def transcribe(file, return_timestamps=False):
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else:
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file_to_transcribe = file
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truncated = False
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else:
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if truncated:
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link="https://github.com/NbAiLab/nostram/blob/main/leverandorer.md"
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@@ -144,13 +160,14 @@ with demo:
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inputs=[
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gr.components.Audio(sources=['upload', 'microphone'], type="filepath"),
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gr.components.Checkbox(label="Inkluder tidsstempler"),
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],
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outputs=gr.HTML(label="text"),
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#outputs="text",
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description=(
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"Transkriber lange lydopptak fra mikrofon eller lydfiler med et enkelt klikk! Demoen bruker den fintunede"
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f" modellen
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),
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allow_flagging="never",
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#show_submit_button=False,
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print(f"Bruker enhet: {device}")
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@spaces.GPU(duration=60 * 2)
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def pipe(file, return_timestamps=False,lang="no"):
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asr = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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text = re.sub(r'(\.{3,}|[.!:?])', lambda m: m.group() + '<br>', text)
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return text
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def transcribe(file, return_timestamps=False,lang="no"):
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waveform, sample_rate = torchaudio.load(file)
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audio_duration = waveform.size(1) / sample_rate
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else:
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file_to_transcribe = file
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truncated = False
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if lang="no":
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if not return_timestamps:
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text = pipe(file_to_transcribe)["text"]
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formatted_text = format_output(text)
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else:
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chunks = pipe(file_to_transcribe, return_timestamps=True)["chunks"]
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text = []
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for chunk in chunks:
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start_time = time.strftime('%H:%M:%S', time.gmtime(chunk["timestamp"][0])) if chunk["timestamp"][0] is not None else "??:??:??"
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end_time = time.strftime('%H:%M:%S', time.gmtime(chunk["timestamp"][1])) if chunk["timestamp"][1] is not None else "??:??:??"
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line = f"[{start_time} -> {end_time}] {chunk['text']}"
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text.append(line)
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formatted_text = "<br>".join(text)
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else:
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if not return_timestamps:
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text = pipe(file_to_transcribe, return_timestams=False,lang="nn")["text"]
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formatted_text = format_output(text)
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else:
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chunks = pipe(file_to_transcribe, return_timestamps=True,lang="nn")["chunks"]
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text = []
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for chunk in chunks:
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start_time = time.strftime('%H:%M:%S', time.gmtime(chunk["timestamp"][0])) if chunk["timestamp"][0] is not None else "??:??:??"
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end_time = time.strftime('%H:%M:%S', time.gmtime(chunk["timestamp"][1])) if chunk["timestamp"][1] is not None else "??:??:??"
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line = f"[{start_time} -> {end_time}] {chunk['text']}"
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text.append(line)
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formatted_text = "<br>".join(text)
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if truncated:
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link="https://github.com/NbAiLab/nostram/blob/main/leverandorer.md"
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inputs=[
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gr.components.Audio(sources=['upload', 'microphone'], type="filepath"),
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gr.components.Checkbox(label="Inkluder tidsstempler"),
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gr.components.Checkbox(label="Nynorsk"),
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],
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outputs=gr.HTML(label="text"),
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#outputs="text",
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description=(
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"Transkriber lange lydopptak fra mikrofon eller lydfiler med et enkelt klikk! <br> Demoen bruker den fintunede"
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f" modellen [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) og 🤗 Transformers til å transkribere lydfiler opp til 30 minutter."
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),
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allow_flagging="never",
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#show_submit_button=False,
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