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Update app.py
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app.py
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@@ -5,19 +5,20 @@ import re
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import random
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import librosa
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import soundfile as sf
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from transformers import pipeline
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from datasets import load_dataset, Audio
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from gradio_client import Client
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from stats_data import get_indomain_breakdown, get_experimental_summary, SPEAKER_META
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# 1. Initialize
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whisper_asr = pipeline(
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"automatic-speech-recognition",
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model="openai/whisper-tiny",
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generate_kwargs={"language": "en", "task": "transcribe"}
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)
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# 2. Configuration
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HF_TOKEN = os.getenv("HF_TOKEN")
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PRIVATE_BACKEND_URL = os.getenv("PRIVATE_BACKEND_URL")
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@@ -26,27 +27,34 @@ def normalize_text(text):
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return re.sub(r'[^\w\s]', '', text).lower().strip()
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def get_sample_logic(speaker_id):
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"""Bypasses internal decoders
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try:
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if
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dataset = load_dataset("resproj007/uaspeech_female", split="train", streaming=True)
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dataset = dataset.cast_column("audio", Audio(decode=False))
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else:
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dataset = load_dataset("abnerh/TORGO-database", split="train", streaming=True)
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dataset = dataset.cast_column("audio", Audio(decode=False))
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def filter_spk(x):
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sid = str(x.get('speaker_id', '')).upper()
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if not sid or sid == "NONE":
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sid = os.path.basename(x['audio']['path']).split('_')[0].upper()
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return sid == speaker_id
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speaker_ds = dataset.filter(filter_spk)
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gt_text = sample.get('transcription') or sample.get('text') or sample.get('sentence') or "Unknown"
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audio_bytes = sample['audio']['bytes']
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audio_data, sample_rate = librosa.load(io.BytesIO(audio_bytes), sr=16000)
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temp_path = "current_sample.wav"
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sf.write(temp_path, audio_data, sample_rate)
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@@ -65,18 +73,15 @@ def run_model_step(audio_path, norm_whisper):
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if not audio_path or not norm_whisper: return "Load data and run Whisper first."
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try:
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client = Client(PRIVATE_BACKEND_URL, hf_token=HF_TOKEN)
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#
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# Adjust api_name to match your private space definition
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prediction = client.predict(audio_path, norm_whisper, api_name="/predict_dsr")
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return prediction
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except Exception as e:
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return f"Backend Error: {e}.
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# UI
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with gr.Blocks(theme=gr.themes.Soft(), title="Torgo DSR Lab") as demo:
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gr.Markdown("# ⚗️ Torgo DSR Lab")
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gr.Markdown("Stepwise evaluation of standard ASR vs. Neural Reconstruction Layer.")
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current_audio_path = gr.State("")
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with gr.Tab("🔬 Laboratory"):
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@@ -103,20 +108,17 @@ with gr.Blocks(theme=gr.themes.Soft(), title="Torgo DSR Lab") as demo:
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gr.Markdown("# 🔬 Performance Evaluation")
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with gr.Row():
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with gr.Column():
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gr.Markdown(""
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Accuracy is the percentage of samples where the **normalized prediction** (lowercase, no punctuation) exactly matches the **normalized ground truth**.
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""")
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with gr.Column():
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gr.Markdown(""
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* **
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""")
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gr.Markdown("---")
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gr.Markdown("## 1. Torgo In-Domain
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gr.DataFrame(get_indomain_breakdown())
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gr.Markdown("## 2. Experimental
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gr.DataFrame(get_experimental_summary())
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# Connectivity
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import random
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import librosa
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import soundfile as sf
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import torch
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from transformers import pipeline
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from datasets import load_dataset, Audio
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from gradio_client import Client
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from stats_data import get_indomain_breakdown, get_experimental_summary, SPEAKER_META
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# 1. Initialize Whisper Tiny (Forced to English)
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whisper_asr = pipeline(
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"automatic-speech-recognition",
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model="openai/whisper-tiny",
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generate_kwargs={"language": "en", "task": "transcribe"}
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)
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# 2. Secret Configuration
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HF_TOKEN = os.getenv("HF_TOKEN")
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PRIVATE_BACKEND_URL = os.getenv("PRIVATE_BACKEND_URL")
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return re.sub(r'[^\w\s]', '', text).lower().strip()
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def get_sample_logic(speaker_id):
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"""Bypasses internal decoders for stability and handles schema differences."""
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try:
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if "UA" in speaker_id:
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# UA-Speech loading (As per your working Colab code)
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dataset = load_dataset("resproj007/uaspeech_female", split="train", streaming=True)
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dataset = dataset.cast_column("audio", Audio(decode=False))
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# For UA Female shard, we pick a random sample directly
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sample = next(iter(dataset.shuffle(buffer_size=50)))
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gt_text = sample.get('text') or sample.get('transcription') or sample.get('sentence')
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else:
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# Torgo loading (Using path-parsing for Speaker IDs)
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dataset = load_dataset("abnerh/TORGO-database", split="train", streaming=True)
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dataset = dataset.cast_column("audio", Audio(decode=False))
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def filter_spk(x):
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sid = str(x.get('speaker_id', '')).upper()
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if not sid or sid == "NONE":
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sid = os.path.basename(x['audio']['path']).split('_')[0].upper()
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return sid == speaker_id
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speaker_ds = dataset.filter(filter_spk)
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sample = next(iter(speaker_ds.shuffle(buffer_size=20)))
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gt_text = sample.get('transcription') or sample.get('text')
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# Manual Decode via librosa (Bypasses torchcodec requirement)
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audio_bytes = sample['audio']['bytes']
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audio_data, sample_rate = librosa.load(io.BytesIO(audio_bytes), sr=16000)
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temp_path = "current_sample.wav"
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sf.write(temp_path, audio_data, sample_rate)
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if not audio_path or not norm_whisper: return "Load data and run Whisper first."
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try:
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client = Client(PRIVATE_BACKEND_URL, hf_token=HF_TOKEN)
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# Call private Gemma model (Backend uses repetition_penalty=3.0)
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prediction = client.predict(audio_path, norm_whisper, api_name="/predict_dsr")
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return prediction
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except Exception as e:
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return f"Backend Error: {e}. Check if Private Space is Awake."
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# UI Construction
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with gr.Blocks(theme=gr.themes.Soft(), title="Torgo DSR Lab") as demo:
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gr.Markdown("# ⚗️ Torgo DSR Lab")
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current_audio_path = gr.State("")
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with gr.Tab("🔬 Laboratory"):
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gr.Markdown("# 🔬 Performance Evaluation")
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with gr.Row():
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with gr.Column():
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gr.Markdown("### 📏 Metric: Exact Match Accuracy")
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gr.Markdown("Accuracy is the percentage of samples where the **normalized prediction** (lowercase, no punctuation) exactly matches the **ground truth**.")
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with gr.Column():
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gr.Markdown("### 🧪 Model Definitions")
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gr.Markdown("* **5K Pure Model:** Trained on real Torgo speech. Optimized for articulatory accuracy.")
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gr.Markdown("* **10K Triple-Mix Model:** Includes synthetic data and anchors. Tested on unseen speakers (LOSO).")
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gr.Markdown("---")
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gr.Markdown("## 1. Torgo In-Domain Breakdown")
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gr.DataFrame(get_indomain_breakdown())
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gr.Markdown("## 2. Experimental Summary")
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gr.DataFrame(get_experimental_summary())
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# Connectivity
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