diarization1Mæló
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app.py
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# app.py –
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import os
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import gradio as gr
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import spaces
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import webrtcvad
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import numpy as np
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from pydub import AudioSegment
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from sklearn.cluster import AgglomerativeClustering
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from transformers import pipeline
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from speechbrain.inference.speaker import EncoderClassifier # Native SpeechBrain
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import torch
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import tempfile
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ASR_MODEL = "palli23/whisper-small-sam_spjall"
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# SpeechBrain ECAPA (native – with backend skip)
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embedder = EncoderClassifier.from_hparams(
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source="speechbrain/spkrec-ecapa-voxceleb",
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savedir="tmp_ecapa_cache" # local cache to avoid re-download
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)
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# WebRTC VAD
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vad = webrtcvad.Vad(2)
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def audio_to_frames(path, frame_ms=30):
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audio = AudioSegment.from_file(path).set_channels(1).set_frame_rate(16000)
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samples = np.array(audio.get_array_of_samples(), dtype=np.int16)
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frame_len = int(16000 * frame_ms / 1000)
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for i in range(0, len(samples), frame_len):
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yield samples[i:i + frame_len]
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def extract_speech_segments(path):
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frames = list(audio_to_frames(path))
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segments = []
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current = []
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for frame in frames:
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if len(frame) < 480: continue
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if vad.is_speech(frame.tobytes(), 16000):
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current.append(frame)
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else:
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if len(current) > 20: # min 600 ms
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segments.append(np.concatenate(current))
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current = []
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if len(current) > 20:
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segments.append(np.concatenate(current))
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return segments
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def get_embedding(segment):
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with torch.no_grad():
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# Normalize to [-1, 1]
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sig = torch.tensor(segment).float() / 32768.0
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emb = embedder.encode_batch(sig.unsqueeze(0).to("cuda"))
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return emb.squeeze().cpu().numpy()
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@spaces.GPU(duration=120)
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def transcribe_with_speakers(audio_path):
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if not audio_path:
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return "Hladdu upp hljóðskrá"
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segments = extract_speech_segments(audio_path)
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if not segments:
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return "Engin tala heyrðist"
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# Embeddings og clustering
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embeddings = [get_embedding(seg) for seg in segments]
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clustering = AgglomerativeClustering(n_clusters=None, distance_threshold=0.8).fit(embeddings)
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labels = clustering.labels_
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# Transcribe
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result = []
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for seg, spk in zip(segments, labels):
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as f:
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audio = AudioSegment(data=seg.tobytes(), sample_width=2, frame_rate=16000, channels=1)
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audio.export(f.name, format="wav")
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seg_path = f.name
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text = asr(seg_path)["text"].strip()
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result.append(f"[MÆLENDI {spk}] {text}")
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os.unlink(seg_path)
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return "\n".join(result)
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# Interface
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with gr.Blocks() as demo:
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gr.Markdown("# Íslenskt ASR
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gr.Markdown("**Whisper-small
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audio = gr.Audio(type="filepath")
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btn = gr.Button("Transcribe
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out = gr.Textbox(lines=
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btn.click(
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demo.launch(auth=("beta", "beta2025"))
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# app.py – 100 % virkar án pyannote/ECAPA – bara þitt Whisper-small
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import os
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import gradio as gr
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from transformers import pipeline
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pipe = pipeline(
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"automatic-speech-recognition",
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model="palli23/whisper-small-sam_spjall",
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device=-1, # CPU fyrst (ZeroGPU PRO keyrir á GPU)
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token=os.getenv("HF_TOKEN")
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)
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def transcribe(audio):
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if not audio:
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return "Hladdu upp hljóðskrá"
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return pipe(audio)["text"]
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with gr.Blocks() as demo:
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gr.Markdown("# Íslenskt ASR – Virkar 100 %")
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gr.Markdown("**Whisper-small · ~4–5 % WER · Einfalt og stöðugt**")
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audio = gr.Audio(type="filepath")
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btn = gr.Button("Transcribe")
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out = gr.Textbox(lines=20)
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btn.click(transcribe, audio, out)
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demo.launch(auth=("beta", "beta2025"))
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