Update README.md

README.md

@@ -10,5 +10,258 @@ pinned: false
 license: mit
 short_description: Qwen2.5-1.5B-Instruct-gkd-demo
 ---
+Great! I’ve reviewed all your files and prepared a **cleaned-up, ready-to-use version of `app.py`** that includes:
 
-Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
+---
+
+## ✅ What’s Fixed & Improved
+
+| Issue | Fix / Enhancement |
+|------|---------------------|
+| 🔁 **Duplicate TTS Block** | Removed duplicate code in `generate_response_and_audio` |
+| ❌ **Incorrect Condition Check** | Replaced unsafe `all([...])` with proper `is not None` checks |
+| 📏 **Long Text Handling (TTS)** | Added chunking to avoid exceeding 512 token limit |
+| 🧠 **LLM Generation Safety** | Ensures `generated_text` is always defined |
+| ⚙️ **Model Loading Optimization** | Moved model loading into the first request (Hugging Face Spaces friendly) |
+| 🧼 **Code Cleanliness** | Better structure, comments, and readability |
+
+---
+
+## 📦 Final Version of `app.py`
+
+Here is your updated file:
+
+```python
+import gradio as gr
+import torch
+from transformers import (
+    AutoTokenizer,
+    AutoModelForCausalLM,
+    SpeechT5Processor,
+    SpeechT5ForTextToSpeech,
+    SpeechT5HifiGan,
+    WhisperProcessor,
+    WhisperForConditionalGeneration
+)
+from datasets import load_dataset
+import os
+import spaces
+import tempfile
+import soundfile as sf
+import librosa
+
+# --- Configuration ---
+HUGGINGFACE_MODEL_ID = "HuggingFaceH4/Qwen2.5-1.5B-Instruct-gkd"
+TORCH_DTYPE = torch.bfloat16
+MAX_NEW_TOKENS = 512
+DO_SAMPLE = True
+TEMPERATURE = 0.7
+TOP_K = 50
+TOP_P = 0.95
+
+TTS_MODEL_ID = "microsoft/speecht5_tts"
+TTS_VOCODER_ID = "microsoft/speecht5_hifigan"
+STT_MODEL_ID = "openai/whisper-small"
+
+# --- Global Variables ---
+tokenizer = None
+llm_model = None
+tts_processor = None
+tts_model = None
+tts_vocoder = None
+speaker_embeddings = None
+whisper_processor = None
+whisper_model = None
+first_load = True
+
+# --- Helper: Split Text Into Chunks ---
+def split_text_into_chunks(text, max_chars=400):
+    sentences = text.replace("...", ".").split(". ")
+    chunks = []
+    current_chunk = ""
+    for sentence in sentences:
+        if len(current_chunk) + len(sentence) + 2 < max_chars:
+            current_chunk += ". " + sentence if current_chunk else sentence
+        else:
+            chunks.append(current_chunk)
+            current_chunk = sentence
+    if current_chunk:
+        chunks.append(current_chunk)
+    return [f"{chunk}." for chunk in chunks if chunk.strip()]
+
+# --- Load Models Function ---
+@spaces.GPU
+def load_models():
+    global tokenizer, llm_model, tts_processor, tts_model, tts_vocoder, speaker_embeddings, whisper_processor, whisper_model
+    hf_token = os.environ.get("HF_TOKEN")
+
+    # LLM
+    if tokenizer is None or llm_model is None:
+        try:
+            tokenizer = AutoTokenizer.from_pretrained(HUGGINGFACE_MODEL_ID, token=hf_token)
+            if tokenizer.pad_token is None:
+                tokenizer.pad_token = tokenizer.eos_token
+            llm_model = AutoModelForCausalLM.from_pretrained(
+                HUGGINGFACE_MODEL_ID,
+                torch_dtype=TORCH_DTYPE,
+                device_map="auto",
+                token=hf_token
+            ).eval()
+            print("LLM loaded successfully.")
+        except Exception as e:
+            print(f"Error loading LLM: {e}")
+
+    # TTS
+    if tts_processor is None or tts_model is None or tts_vocoder is None:
+        try:
+            tts_processor = SpeechT5Processor.from_pretrained(TTS_MODEL_ID, token=hf_token)
+            tts_model = SpeechT5ForTextToSpeech.from_pretrained(TTS_MODEL_ID, token=hf_token)
+            tts_vocoder = SpeechT5HifiGan.from_pretrained(TTS_VOCODER_ID, token=hf_token)
+            embeddings = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation", token=hf_token)
+            speaker_embeddings = torch.tensor(embeddings[7306]["xvector"]).unsqueeze(0)
+            device = llm_model.device if llm_model else 'cpu'
+            tts_model.to(device)
+            tts_vocoder.to(device)
+            speaker_embeddings = speaker_embeddings.to(device)
+            print("TTS models loaded.")
+        except Exception as e:
+            print(f"Error loading TTS: {e}")
+
+    # STT
+    if whisper_processor is None or whisper_model is None:
+        try:
+            whisper_processor = WhisperProcessor.from_pretrained(STT_MODEL_ID, token=hf_token)
+            whisper_model = WhisperForConditionalGeneration.from_pretrained(STT_MODEL_ID, token=hf_token)
+            device = llm_model.device if llm_model else 'cpu'
+            whisper_model.to(device)
+            print("Whisper loaded.")
+        except Exception as e:
+            print(f"Error loading Whisper: {e}")
+
+# --- Generate Response and Audio ---
+@spaces.GPU
+def generate_response_and_audio(message, history):
+    global first_load
+    if first_load:
+        load_models()
+        first_load = False
+
+    global tokenizer, llm_model, tts_processor, tts_model, tts_vocoder, speaker_embeddings
+
+    if tokenizer is None or llm_model is None:
+        return [{"role": "assistant", "content": "Error: LLM not loaded."}], None
+
+    messages = history.copy()
+    messages.append({"role": "user", "content": message})
+
+    try:
+        input_text = tokenizer.apply_chat_template(messages, tokenize=False, add_generation_prompt=True)
+    except:
+        input_text = ""
+        for item in history:
+            input_text += f"{item['role'].capitalize()}: {item['content']}\n"
+        input_text += f"User: {message}\nAssistant:"
+
+    try:
+        inputs = tokenizer(input_text, return_tensors="pt", padding=True, truncation=True).to(llm_model.device)
+        output_ids = llm_model.generate(
+            inputs["input_ids"],
+            attention_mask=inputs["attention_mask"],
+            max_new_tokens=MAX_NEW_TOKENS,
+            do_sample=DO_SAMPLE,
+            temperature=TEMPERATURE,
+            top_k=TOP_K,
+            top_p=TOP_P,
+            pad_token_id=tokenizer.eos_token_id
+        )
+        generated_text = tokenizer.decode(output_ids[0][inputs["input_ids"].shape[-1]:], skip_special_tokens=True).strip()
+    except Exception as e:
+        print(f"LLM error: {e}")
+        return history + [{"role": "assistant", "content": "I had an issue generating a response."}], None
+
+    audio_path = None
+    if None not in [tts_processor, tts_model, tts_vocoder, speaker_embeddings]:
+        try:
+            device = llm_model.device
+            text_chunks = split_text_into_chunks(generated_text)
+
+            full_speech = []
+            for chunk in text_chunks:
+                tts_inputs = tts_processor(text=chunk, return_tensors="pt", max_length=512, truncation=True).to(device)
+                speech = tts_model.generate_speech(tts_inputs["input_ids"], speaker_embeddings, vocoder=tts_vocoder)
+                full_speech.append(speech.cpu())
+
+            full_speech_tensor = torch.cat(full_speech, dim=0)
+
+            with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp_file:
+                audio_path = tmp_file.name
+                sf.write(audio_path, full_speech_tensor.numpy(), samplerate=16000)
+
+        except Exception as e:
+            print(f"TTS error: {e}")
+
+    return history + [{"role": "assistant", "content": generated_text}], audio_path
+
+# --- Transcribe Audio ---
+@spaces.GPU
+def transcribe_audio(filepath):
+    global first_load
+    if first_load:
+        load_models()
+        first_load = False
+
+    global whisper_processor, whisper_model
+    if whisper_model is None:
+        return "Whisper model not loaded."
+
+    try:
+        audio, sr = librosa.load(filepath, sr=16000)
+        inputs = whisper_processor(audio, sampling_rate=sr, return_tensors="pt").input_features.to(whisper_model.device)
+        outputs = whisper_model.generate(inputs)
+        return whisper_processor.batch_decode(outputs, skip_special_tokens=True)[0]
+    except Exception as e:
+        return f"Transcription failed: {e}"
+
+# --- Gradio UI ---
+with gr.Blocks() as demo:
+    gr.Markdown("# Qwen2.5 Chatbot with Voice Input/Output")
+    
+    with gr.Tab("Chat"):
+        chatbot = gr.Chatbot(type='messages')
+        text_input = gr.Textbox(placeholder="Type your message...")
+        audio_output = gr.Audio(label="Response Audio", autoplay=True)
+        text_input.submit(generate_response_and_audio, [text_input, chatbot], [chatbot, audio_output])
+
+    with gr.Tab("Transcribe"):
+        audio_input = gr.Audio(type="filepath", label="Upload Audio")
+        transcribed = gr.Textbox(label="Transcription")
+        audio_input.upload(transcribe_audio, audio_input, transcribed)
+
+    clear_btn = gr.Button("Clear All")
+    clear_btn.click(lambda: ([], "", None), None, [chatbot, text_input, audio_output])
+
+demo.queue().launch()
+```
+
+---
+
+## ✅ Instructions for Uploading to Hugging Face Spaces
+
+1. **Go to your Space**: https://huggingface.co/spaces/ajsbsd/Qwen2.5-1.5B-Instruct-gkd-demo
+2. **Pause the CI**: Go to `Settings > Runtime`, and switch from "Always On" to "Manual"
+3. **Delete old app.py**
+4. **Upload this new file** as `app.py`
+5. **Start the CI again**
+
+---
+
+## 🧩 Optional Enhancements
+
+Would you like me to help you with any of the following?
+
+- Add **status indicators** during model loading or generation
+- Allow **microphone input** directly in chat tab
+- Use `gr.State()` to store chat history more efficiently
+- Package models into a custom repo for faster load times
+
+Just let me know what you'd like next!Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference