Update app.py
Browse files
app.py
CHANGED
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import gradio as gr
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import tempfile
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import numpy as np
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from faster_whisper import WhisperModel
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from scipy.io.wavfile import write
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model = WhisperModel(
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"
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device="cpu",
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compute_type="int8"
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)
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def transcribe(audio):
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if audio is None:
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return
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sr,
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f.name,
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language="en"
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)
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if
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return
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with gr.Blocks() as demo:
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@@ -44,20 +64,21 @@ with gr.Blocks() as demo:
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gr.Markdown("# Real-Time English Speech Recognition")
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audio = gr.Audio(
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streaming=True,
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sources=["microphone"],
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type="numpy"
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)
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label="
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lines=
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)
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audio.stream(
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transcribe,
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audio,
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)
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demo.launch()
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import gradio as gr
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import numpy as np
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import soundfile as sf
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import tempfile
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from faster_whisper import WhisperModel
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# Fastest practical model
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model = WhisperModel(
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"turbo",
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device="cpu",
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compute_type="int8"
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)
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full_transcript = ""
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last_segment = ""
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def transcribe(audio):
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global full_transcript
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global last_segment
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if audio is None:
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return full_transcript
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sr, data = audio
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if len(data) < sr // 2:
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return full_transcript
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with tempfile.NamedTemporaryFile(suffix=".wav") as f:
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sf.write(f.name, data, sr)
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segments, info = model.transcribe(
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f.name,
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language="en",
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vad_filter=True,
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beam_size=1,
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best_of=1,
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temperature=0
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)
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current_text = " ".join(
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segment.text.strip()
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for segment in segments
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)
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if not current_text:
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return full_transcript
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if current_text != last_segment:
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if full_transcript:
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full_transcript += " "
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full_transcript += current_text
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last_segment = current_text
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return full_transcript
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with gr.Blocks() as demo:
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gr.Markdown("# Real-Time English Speech Recognition")
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audio = gr.Audio(
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sources=["microphone"],
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streaming=True,
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type="numpy"
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)
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output = gr.Textbox(
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label="Transcript",
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lines=12
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)
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audio.stream(
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fn=transcribe,
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inputs=audio,
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outputs=output,
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stream_every=1
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)
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demo.launch()
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