diff --git a/README.md b/README.md
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@@ -1,336 +1,372 @@
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- English |
- 简体中文 |
- 繁體中文 |
- 한국어 |
- Español |
- 日本語 |
- हिन्दी |
- Русский |
- Português |
- తెలుగు |
- Français |
- Deutsch |
- Italiano |
- Tiếng Việt |
- العربية |
- اردو |
- বাংলা |
-
-
-
-
- State-of-the-art pretrained models for inference and training
-
-
-
-
-
-Transformers acts as the model-definition framework for state-of-the-art machine learning models in text, computer
-vision, audio, video, and multimodal model, for both inference and training.
-
-It centralizes the model definition so that this definition is agreed upon across the ecosystem. `transformers` is the
-pivot across frameworks: if a model definition is supported, it will be compatible with the majority of training
-frameworks (Axolotl, Unsloth, DeepSpeed, FSDP, PyTorch-Lightning, ...), inference engines (vLLM, SGLang, TGI, ...),
-and adjacent modeling libraries (llama.cpp, mlx, ...) which leverage the model definition from `transformers`.
-
-We pledge to help support new state-of-the-art models and democratize their usage by having their model definition be
-simple, customizable, and efficient.
-
-There are over 1M+ Transformers [model checkpoints](https://huggingface.co/models?library=transformers&sort=trending) on the [Hugging Face Hub](https://huggingface.com/models) you can use.
-
-Explore the [Hub](https://huggingface.com/) today to find a model and use Transformers to help you get started right away.
-
-## Installation
-
-Transformers works with Python 3.9+, and [PyTorch](https://pytorch.org/get-started/locally/) 2.1+.
-
-Create and activate a virtual environment with [venv](https://docs.python.org/3/library/venv.html) or [uv](https://docs.astral.sh/uv/), a fast Rust-based Python package and project manager.
-
-```py
-# venv
-python -m venv .my-env
-source .my-env/bin/activate
-# uv
-uv venv .my-env
-source .my-env/bin/activate
+# Isan Pin AI - Traditional Thai Music Generation System
+
+A comprehensive AI system for generating authentic traditional Isan Pin music from Northeast Thailand using deep learning and cultural preservation techniques.
+
+## 🎵 Overview
+
+Isan Pin AI is a sophisticated system that combines:
+- **Cultural Preservation**: Maintains authentic Thai musical traditions
+- **Advanced AI**: Uses Facebook MusicGen fine-tuned on traditional Isan Pin music
+- **Style Classification**: CNN-based recognition of 5 traditional styles
+- **Quality Assessment**: Multi-dimensional evaluation with cultural authenticity
+- **User-Friendly Interface**: Both command-line and web interfaces
+
+## 🎯 Key Features
+
+### Traditional Music Styles Supported
+- **Lam Plearn** (หมอลำเพลิน) - Slow, contemplative style (60-90 BPM)
+- **Lam Sing** (หมอลำซิ่ง) - Fast, energetic style (120-160 BPM)
+- **Lam Klorn** (หมอลำกลอน) - Moderate, poetic style (80-120 BPM)
+- **Lam Tad** (หมอลำตัด) - Medium-fast, narrative style (100-140 BPM)
+- **Lam Puen** (หมอลำปึน) - Moderate, storytelling style (90-130 BPM)
+
+### AI Capabilities
+- 🎼 **Text-to-Music Generation**: Create music from Thai/English descriptions
+- 🔄 **Style Transfer**: Convert between different Isan Pin styles
+- 🔍 **Style Classification**: Automatically identify musical styles
+- 📊 **Quality Assessment**: Evaluate cultural authenticity and audio quality
+- 🌐 **Web Interface**: User-friendly Gradio and FastAPI interfaces
+
+## 🚀 Quick Start
+
+### Installation
+
+```bash
+# Install dependencies
+pip install -r requirements.txt
+
+# Or install as package
+pip install -e .
```
-Install Transformers in your virtual environment.
+### Basic Usage
+
+```bash
+# Generate traditional Isan Pin music
+python main.py generate --prompt "สร้างเสียงพิณแบบหมอลำเพลินช้าๆ เศร้าๆ" --style lam_plearn --duration 30
+
+# Launch web interface
+python main.py web --port 8000
-```py
-# pip
-pip install "transformers[torch]"
+# Train classifier on your data
+python main.py train-classifier --data-dir ./dataset --output-dir ./models
-# uv
-uv pip install "transformers[torch]"
+# Evaluate generated music
+python main.py evaluate --audio-file ./generated_music.wav --reference-style lam_plearn
```
-Install Transformers from source if you want the latest changes in the library or are interested in contributing. However, the *latest* version may not be stable. Feel free to open an [issue](https://github.com/huggingface/transformers/issues) if you encounter an error.
+### Python API
-```shell
-git clone https://github.com/huggingface/transformers.git
-cd transformers
+```python
+from src import IsanPinInference, IsanPinEvaluator
-# pip
-pip install '.[torch]'
+# Initialize inference system
+inference = IsanPinInference(
+ classifier_path="./models/classifier.pth",
+ generator_path="./models/generator.pth"
+)
+
+# Generate music
+results = inference.generate_music(
+ description="Traditional Isan Pin music with slow tempo and contemplative mood",
+ style="lam_plearn",
+ duration=30.0,
+ num_samples=1
+)
-# uv
-uv pip install '.[torch]'
+# Evaluate quality
+evaluator = IsanPinEvaluator()
+results = evaluator.evaluate_single_audio(
+ audio="./generated_music.wav",
+ reference_style="lam_plearn"
+)
+print(f"Cultural Authenticity Score: {results['cultural_authenticity']['cultural_authenticity_score']}")
```
-## Quickstart
+## 🎨 Web Interface
+
+Access the web interface at `http://localhost:8000` after running:
-Get started with Transformers right away with the [Pipeline](https://huggingface.co/docs/transformers/pipeline_tutorial) API. The `Pipeline` is a high-level inference class that supports text, audio, vision, and multimodal tasks. It handles preprocessing the input and returns the appropriate output.
+```bash
+python main.py web --port 8000
+```
-Instantiate a pipeline and specify model to use for text generation. The model is downloaded and cached so you can easily reuse it again. Finally, pass some text to prompt the model.
+### Features
+- **Generate Music**: Create music from text descriptions
+- **Style Transfer**: Apply different Isan Pin styles to existing audio
+- **Evaluate Audio**: Assess quality and cultural authenticity
+- **Download Results**: Export generated music in various formats
-```py
-from transformers import pipeline
+## 📁 Project Structure
-pipeline = pipeline(task="text-generation", model="Qwen/Qwen2.5-1.5B")
-pipeline("the secret to baking a really good cake is ")
-[{'generated_text': 'the secret to baking a really good cake is 1) to use the right ingredients and 2) to follow the recipe exactly. the recipe for the cake is as follows: 1 cup of sugar, 1 cup of flour, 1 cup of milk, 1 cup of butter, 1 cup of eggs, 1 cup of chocolate chips. if you want to make 2 cakes, how much sugar do you need? To make 2 cakes, you will need 2 cups of sugar.'}]
+```
+isan-pin-ai/
+├── src/ # Source code
+│ ├── config.py # Configuration settings
+│ ├── data/ # Data processing modules
+│ │ ├── collection.py # Audio collection and classification
+│ │ └── preprocessing.py # Audio preprocessing and dataset creation
+│ ├── models/ # AI models
+│ │ ├── classification.py # CNN-based style classifier
+│ │ └── musicgen.py # MusicGen fine-tuning
+│ ├── inference/ # Inference pipeline
+│ │ └── inference.py # High-level inference system
+│ ├── evaluation/ # Quality assessment
+│ │ └── evaluator.py # Comprehensive evaluation system
+│ └── web/ # Web interfaces
+│ └── app.py # FastAPI + Gradio application
+├── main.py # Command-line interface
+├── requirements.txt # Dependencies
+└── README.md # This file
```
-To chat with a model, the usage pattern is the same. The only difference is you need to construct a chat history (the input to `Pipeline`) between you and the system.
+## 🎯 Detailed Usage Examples
-> [!TIP]
-> You can also chat with a model directly from the command line.
-> ```shell
-> transformers chat Qwen/Qwen2.5-0.5B-Instruct
-> ```
+### 1. Data Collection and Preprocessing
-```py
-import torch
-from transformers import pipeline
+```bash
+# Collect and classify raw audio files
+python main.py collect-data \
+ --input-dir ./raw_audio \
+ --output-dir ./processed \
+ --sample-rate 48000 \
+ --classify
+```
-chat = [
- {"role": "system", "content": "You are a sassy, wise-cracking robot as imagined by Hollywood circa 1986."},
- {"role": "user", "content": "Hey, can you tell me any fun things to do in New York?"}
-]
+### 2. Training Models
+
+```bash
+# Train style classifier
+python main.py train-classifier \
+ --data-dir ./dataset \
+ --output-dir ./models/classifier \
+ --epochs 50 \
+ --batch-size 16
+
+# Fine-tune music generator
+python main.py train-generator \
+ --data-dir ./dataset \
+ --output-dir ./models/generator \
+ --base-model facebook/musicgen-small \
+ --epochs 10 \
+ --batch-size 4
+```
-pipeline = pipeline(task="text-generation", model="meta-llama/Meta-Llama-3-8B-Instruct", dtype=torch.bfloat16, device_map="auto")
-response = pipeline(chat, max_new_tokens=512)
-print(response[0]["generated_text"][-1]["content"])
+### 3. Music Generation
+
+```bash
+# Generate with specific style
+python main.py generate \
+ --prompt "Traditional Isan Pin music for meditation and relaxation" \
+ --style lam_plearn \
+ --duration 60 \
+ --output-file ./meditation_pin.wav
+
+# Generate with mood
+python main.py generate \
+ --prompt "เสียงพิณแบบไทยดั้งเดิม ฟังเพลินๆ" \
+ --style lam_sing \
+ --mood joyful \
+ --duration 45
```
-Expand the examples below to see how `Pipeline` works for different modalities and tasks.
+### 4. Style Transfer
+
+```bash
+# Transfer to different style
+python main.py transfer-style \
+ --input-audio ./input_music.wav \
+ --target-style lam_klorn \
+ --output-file ./transferred_klorn.wav \
+ --strength 0.8
+```
-
-Automatic speech recognition
+### 5. Evaluation
-```py
-from transformers import pipeline
+```bash
+# Evaluate single file
+python main.py evaluate \
+ --audio-file ./generated_music.wav \
+ --reference-style lam_plearn \
+ --detailed
-pipeline = pipeline(task="automatic-speech-recognition", model="openai/whisper-large-v3")
-pipeline("https://huggingface.co/datasets/Narsil/asr_dummy/resolve/main/mlk.flac")
-{'text': ' I have a dream that one day this nation will rise up and live out the true meaning of its creed.'}
+# Evaluate dataset
+python main.py evaluate-dataset \
+ --data-dir ./test_dataset \
+ --output-report ./evaluation_report.json
```
-
+## 🔬 Evaluation Metrics
-
-Image classification
+The system provides comprehensive evaluation:
-
-
+### Basic Quality Metrics
+- **RMS Energy**: Audio signal strength
+- **Dynamic Range**: Variation in loudness
+- **Zero Crossing Rate**: Rhythmic complexity
+- **Quality Score**: Overall technical quality (0-1)
-```py
-from transformers import pipeline
+### Style Consistency
+- **Tempo Consistency**: Alignment with style-specific tempo ranges
+- **Spectral Consistency**: Matching spectral characteristics
+- **MFCC Consistency**: Mel-frequency cepstral coefficients alignment
-pipeline = pipeline(task="image-classification", model="facebook/dinov2-small-imagenet1k-1-layer")
-pipeline("https://huggingface.co/datasets/Narsil/image_dummy/raw/main/parrots.png")
-[{'label': 'macaw', 'score': 0.997848391532898},
- {'label': 'sulphur-crested cockatoo, Kakatoe galerita, Cacatua galerita',
- 'score': 0.0016551691805943847},
- {'label': 'lorikeet', 'score': 0.00018523589824326336},
- {'label': 'African grey, African gray, Psittacus erithacus',
- 'score': 7.85409429227002e-05},
- {'label': 'quail', 'score': 5.502637941390276e-05}]
+### Cultural Authenticity
+- **Tempo Authenticity**: Traditional tempo ranges
+- **Rhythmic Authenticity**: Cultural rhythmic patterns
+- **Spectral Authenticity**: Traditional timbral characteristics
+
+### Reference Comparison
+- **Similarity Score**: Comparison with reference Isan Pin recordings
+- **Classification Consistency**: Agreement with style classifier
+
+## 🎛️ Configuration
+
+Edit `src/config.py` to customize:
+
+```python
+# Audio settings
+AUDIO_CONFIG = {
+ "sample_rate": 48000,
+ "n_fft": 2048,
+ "hop_length": 512,
+ "n_mels": 128,
+}
+
+# Style definitions
+STYLE_DEFINITIONS = {
+ "lam_plearn": {
+ "thai_name": "หมอลำเพลิน",
+ "tempo_range": [60, 90],
+ "mood": ["sad", "romantic", "contemplative"],
+ }
+ # ... more styles
+}
```
-
+## 🌐 API Usage
-
-Visual question answering
+### FastAPI Endpoints
-
-
+```bash
+# Get available styles
+curl http://localhost:8000/styles
-```py
-from transformers import pipeline
+# Generate music
+curl -X POST http://localhost:8000/generate \
+ -H "Content-Type: application/json" \
+ -d '{
+ "description": "Traditional Isan Pin music",
+ "style": "lam_plearn",
+ "duration": 30,
+ "num_samples": 1
+ }'
-pipeline = pipeline(task="visual-question-answering", model="Salesforce/blip-vqa-base")
-pipeline(
- image="https://huggingface.co/datasets/huggingface/documentation-images/resolve/main/transformers/tasks/idefics-few-shot.jpg",
- question="What is in the image?",
-)
-[{'answer': 'statue of liberty'}]
+# Upload and evaluate
+curl -X POST http://localhost:8000/evaluate \
+ -F "audio_file=@./music.wav" \
+ -F "reference_style=lam_plearn"
```
-
+### Gradio Interface
-## Why should I use Transformers?
+Access the user-friendly interface at `http://localhost:8000/gradio`
-1. Easy-to-use state-of-the-art models:
- - High performance on natural language understanding & generation, computer vision, audio, video, and multimodal tasks.
- - Low barrier to entry for researchers, engineers, and developers.
- - Few user-facing abstractions with just three classes to learn.
- - A unified API for using all our pretrained models.
+## 🧪 Advanced Features
-1. Lower compute costs, smaller carbon footprint:
- - Share trained models instead of training from scratch.
- - Reduce compute time and production costs.
- - Dozens of model architectures with 1M+ pretrained checkpoints across all modalities.
+### Batch Processing
-1. Choose the right framework for every part of a models lifetime:
- - Train state-of-the-art models in 3 lines of code.
- - Move a single model between PyTorch/JAX/TF2.0 frameworks at will.
- - Pick the right framework for training, evaluation, and production.
+```python
+# Generate multiple samples
+results = inference.batch_generate(
+ descriptions=["slow meditation music", "fast dance music"],
+ styles=["lam_plearn", "lam_sing"],
+ num_samples_per_description=3,
+ max_workers=4
+)
+```
-1. Easily customize a model or an example to your needs:
- - We provide examples for each architecture to reproduce the results published by its original authors.
- - Model internals are exposed as consistently as possible.
- - Model files can be used independently of the library for quick experiments.
-
-
-
-Audio
-
-- Audio classification with [Whisper](https://huggingface.co/openai/whisper-large-v3-turbo)
-- Automatic speech recognition with [Moonshine](https://huggingface.co/UsefulSensors/moonshine)
-- Keyword spotting with [Wav2Vec2](https://huggingface.co/superb/wav2vec2-base-superb-ks)
-- Speech to speech generation with [Moshi](https://huggingface.co/kyutai/moshiko-pytorch-bf16)
-- Text to audio with [MusicGen](https://huggingface.co/facebook/musicgen-large)
-- Text to speech with [Bark](https://huggingface.co/suno/bark)
-
-
-
-
-Computer vision
-
-- Automatic mask generation with [SAM](https://huggingface.co/facebook/sam-vit-base)
-- Depth estimation with [DepthPro](https://huggingface.co/apple/DepthPro-hf)
-- Image classification with [DINO v2](https://huggingface.co/facebook/dinov2-base)
-- Keypoint detection with [SuperPoint](https://huggingface.co/magic-leap-community/superpoint)
-- Keypoint matching with [SuperGlue](https://huggingface.co/magic-leap-community/superglue_outdoor)
-- Object detection with [RT-DETRv2](https://huggingface.co/PekingU/rtdetr_v2_r50vd)
-- Pose Estimation with [VitPose](https://huggingface.co/usyd-community/vitpose-base-simple)
-- Universal segmentation with [OneFormer](https://huggingface.co/shi-labs/oneformer_ade20k_swin_large)
-- Video classification with [VideoMAE](https://huggingface.co/MCG-NJU/videomae-large)
-
-
-
-
-Multimodal
-
-- Audio or text to text with [Qwen2-Audio](https://huggingface.co/Qwen/Qwen2-Audio-7B)
-- Document question answering with [LayoutLMv3](https://huggingface.co/microsoft/layoutlmv3-base)
-- Image or text to text with [Qwen-VL](https://huggingface.co/Qwen/Qwen2.5-VL-3B-Instruct)
-- Image captioning [BLIP-2](https://huggingface.co/Salesforce/blip2-opt-2.7b)
-- OCR-based document understanding with [GOT-OCR2](https://huggingface.co/stepfun-ai/GOT-OCR-2.0-hf)
-- Table question answering with [TAPAS](https://huggingface.co/google/tapas-base)
-- Unified multimodal understanding and generation with [Emu3](https://huggingface.co/BAAI/Emu3-Gen)
-- Vision to text with [Llava-OneVision](https://huggingface.co/llava-hf/llava-onevision-qwen2-0.5b-ov-hf)
-- Visual question answering with [Llava](https://huggingface.co/llava-hf/llava-1.5-7b-hf)
-- Visual referring expression segmentation with [Kosmos-2](https://huggingface.co/microsoft/kosmos-2-patch14-224)
-
-
-
-
-NLP
-
-- Masked word completion with [ModernBERT](https://huggingface.co/answerdotai/ModernBERT-base)
-- Named entity recognition with [Gemma](https://huggingface.co/google/gemma-2-2b)
-- Question answering with [Mixtral](https://huggingface.co/mistralai/Mixtral-8x7B-v0.1)
-- Summarization with [BART](https://huggingface.co/facebook/bart-large-cnn)
-- Translation with [T5](https://huggingface.co/google-t5/t5-base)
-- Text generation with [Llama](https://huggingface.co/meta-llama/Llama-3.2-1B)
-- Text classification with [Qwen](https://huggingface.co/Qwen/Qwen2.5-0.5B)
-
-
-
-## Citation
-
-We now have a [paper](https://www.aclweb.org/anthology/2020.emnlp-demos.6/) you can cite for the 🤗 Transformers library:
-```bibtex
-@inproceedings{wolf-etal-2020-transformers,
- title = "Transformers: State-of-the-Art Natural Language Processing",
- author = "Thomas Wolf and Lysandre Debut and Victor Sanh and Julien Chaumond and Clement Delangue and Anthony Moi and Pierric Cistac and Tim Rault and Rémi Louf and Morgan Funtowicz and Joe Davison and Sam Shleifer and Patrick von Platen and Clara Ma and Yacine Jernite and Julien Plu and Canwen Xu and Teven Le Scao and Sylvain Gugger and Mariama Drame and Quentin Lhoest and Alexander M. Rush",
- booktitle = "Proceedings of the 2020 Conference on Empirical Methods in Natural Language Processing: System Demonstrations",
- month = oct,
- year = "2020",
- address = "Online",
- publisher = "Association for Computational Linguistics",
- url = "https://www.aclweb.org/anthology/2020.emnlp-demos.6",
- pages = "38--45"
-}
+### Style Interpolation
+
+```python
+# Create smooth transitions between styles
+interpolations = inference.interpolate_styles(
+ description="Traditional Isan Pin music",
+ style1="lam_plearn",
+ style2="lam_sing",
+ num_steps=5,
+ duration=30.0
+)
```
+
+### Custom Training
+
+```python
+# Prepare dataset
+dataset = IsanPinDataset(
+ audio_paths=audio_files,
+ descriptions=descriptions,
+ preprocessor=AudioPreprocessor()
+)
+
+# Train classifier
+classifier = IsanPinClassifier()
+history = classifier.train(
+ train_dataset=train_dataset,
+ val_dataset=val_dataset,
+ num_epochs=50,
+ save_path="./models/classifier.pth"
+)
+```
+
+## 🔧 Troubleshooting
+
+### Common Issues
+
+1. **CUDA Out of Memory**: Reduce batch size or use CPU
+2. **Audio Quality Issues**: Adjust temperature and guidance_scale parameters
+3. **Style Misclassification**: Retrain classifier with more diverse data
+4. **Slow Generation**: Use smaller model or reduce duration
+
+### Performance Tips
+
+- Use GPU for faster training and generation
+- Batch process multiple files when possible
+- Pre-compute reference features for faster evaluation
+- Cache generated samples for repeated use
+
+## 📈 Future Enhancements
+
+- [ ] Real-time generation capabilities
+- [ ] Mobile app development
+- [ ] Integration with Thai music notation
+- [ ] Collaborative filtering for recommendations
+- [ ] Multi-instrument generation
+- [ ] Live performance integration
+
+## 🤝 Contributing
+
+1. Fork the repository
+2. Create feature branch (`git checkout -b feature/amazing-feature`)
+3. Commit changes (`git commit -m 'Add amazing feature'`)
+4. Push to branch (`git push origin feature/amazing-feature`)
+5. Open a Pull Request
+
+## 📄 License
+
+This project is licensed under the MIT License - see the [LICENSE](LICENSE) file for details.
+
+## 🙏 Acknowledgments
+
+- Facebook MusicGen team for the base model
+- Thai musicologists and cultural experts
+- Isan Pin musicians for preserving traditional music
+- Open source community for supporting tools
+
+## 📞 Contact
+
+For questions or support, please open an issue on GitHub or contact the development team.
+
+---
+
+**Preserve Thai Culture Through AI** 🎵🇹🇭
\ No newline at end of file
diff --git a/README_PIN_AI_PROMPTS.md b/README_PIN_AI_PROMPTS.md
new file mode 100644
index 000000000000..13f0976ef3cd
--- /dev/null
+++ b/README_PIN_AI_PROMPTS.md
@@ -0,0 +1,476 @@
+# พิณAI - Isan Pin AI Prompts Collection
+
+## ภาษาไทย | [English](#english-version)
+
+คอลเล็กชันของ prompts สำหรับการพัฒนา AI สร้างเสียงพิณอีสาน รวมถึงการเก็บข้อมูล, การสอนโมเดล, และการสร้างเสียง
+
+## 📋 สารบัญ
+
+1. [การเก็บและจัดการข้อมูล](#การเก็บและจัดการข้อมูล)
+2. [การสอนโมเดล](#การสอนโมเดล)
+3. [การสร้างเสียง](#การสร้างเสียง)
+4. [การถ่ายโอนสไตล์](#การถ่ายโอนสไตล์)
+5. [การประเมินผล](#การประเมินผล)
+6. [การพัฒนาแอปพลิเคชัน](#การพัฒนาแอปพลิเคชัน)
+7. [การวิจัยและปรับปรุง](#การวิจัยและปรับปรุง)
+
+## 🔧 การเก็บและจัดการข้อมูล
+
+### การจัดหมวดหมู่ไฟล์เสียง
+
+```python
+from pin_ai_prompts import DATA_COLLECTION_PROMPTS
+
+classification_prompt = DATA_COLLECTION_PROMPTS["audio_classification"]
+print(classification_prompt)
+```
+
+ระบุรายละเอียดสำหรับการวิเคราะห์ไฟล์เสียง:
+- สไตล์การเล่น: หมอลำเพลิน, หมอลำซิ่ง, หมอลำกลอน, ลำตัด, ลำพื้น
+- Tempo: ช้า (60-80 BPM), ปานกลาง (80-120 BPM), เร็ว (120-160 BPM)
+- อารมณ์: เศร้า, สนุกสนาน, โรแมนติก, ตื่นเต้น, เฉยๆ
+- ภูมิภาค: อีสานเหนือ, อีสานใต้, อีสานกลาง
+- เทคนิคพิเศษ: การสั่น, การเล่นเร็ว, การเด้ง, การประสาน
+
+### การสร้าง Dataset
+
+```python
+dataset_prompt = DATA_COLLECTION_PROMPTS["dataset_creation"]
+```
+
+สร้าง training dataset ที่มี:
+- จำนวนตัวอย่าง: 1000 ไฟล์
+- ความยาว: 30-180 วินาที
+- Sample rate: 48kHz, 24-bit
+
+## 🎯 การสอนโมเดล
+
+### Fine-tuning MusicGen
+
+```python
+from pin_ai_prompts import TRAINING_PROMPTS
+
+finetune_config = TRAINING_PROMPTS["musicgen_finetune"]
+print(finetune_config)
+```
+
+รายละเอียดการตั้งค่า:
+- Base Model: facebook/musicgen-small
+- Learning Rate: 5e-5
+- Batch Size: 4
+- Epochs: 50
+- Dataset: 800 training, 150 validation, 50 test
+
+### Training Script
+
+```python
+training_script = TRAINING_PROMPTS["training_script"]
+```
+
+สคริปต์สำหรับการสอนที่รวม:
+- Multi-GPU training
+- Checkpoint management
+- Weights & Biases logging
+- Auto-evaluation
+- Mixed precision training
+
+## 🎵 การสร้างเสียง
+
+### สร้างเสียงใหม่
+
+```python
+from pin_ai_prompts import INFERENCE_PROMPTS
+
+generation_prompt = INFERENCE_PROMPTS["generation"]
+```
+
+รองรับคำสั่งต่างๆ:
+- "สร้างเสียงพิณแบบหมอลำเพลินช้าๆ เศร้าๆ นาน 30 วินาที"
+- "เล่นพิณสไตล์อีสานเหนือ จังหวะเร็ว สนุกสนาน"
+- "ทำเสียงพิณโรแมนติก เหมาะกับงานแต่งงาน"
+
+### Interactive Generation
+
+```python
+interactive_system = INFERENCE_PROMPTS["interactive"]
+```
+
+ระบบสนทนาแบบโต้ตอบ:
+- Real-time generation
+- Progressive refinement
+- User feedback loop
+- Style mixing
+- Variation generation
+
+## 🔄 การถ่ายโอนสไตล์
+
+```python
+from pin_ai_prompts import STYLE_TRANSFER_PROMPTS
+
+transfer_system = STYLE_TRANSFER_PROMPTS["transfer_system"]
+```
+
+การแปลงสไตล์เสียงพิณ:
+- หมอลำเพลิน → หมอลำซิ่ง
+- ช้า → เร็ว
+- เศร้า → สนุกสนาน
+- อีสานเหนือ → อีสานใต้
+
+## 📊 การประเมินผล
+
+### ประเมินคุณภาพเสียง
+
+```python
+from pin_ai_prompts import EVALUATION_PROMPTS
+
+quality_assessment = EVALUATION_PROMPTS["quality_assessment"]
+```
+
+ประเมินตามมิติต่างๆ:
+- Technical Quality (0-10)
+- Musical Quality (0-10)
+- Style Authenticity (0-10)
+- Emotional Impact (0-10)
+
+### วิเคราะห์ประสิทธิภาพโมเดล
+
+```python
+performance_analysis = EVALUATION_PROMPTS["performance_analysis"]
+```
+
+การวิเคราะห์:
+- Generation Quality vs Training Data Size
+- Style Coverage Analysis
+- Tempo and Key Accuracy
+- Diversity Analysis
+- Failure Cases
+
+## 📱 การพัฒนาแอปพลิเคชัน
+
+### Web Application
+
+```python
+from pin_ai_prompts import APP_PROMPTS
+
+web_app_design = APP_PROMPTS["web_app"]
+```
+
+แอปพลิเคชัน "พิณAI - Isan Pin Generator" รวม:
+- Homepage และ Quick generation
+- Advanced settings
+- Style transfer
+- Gallery และ community features
+- Learning center
+
+### Mobile Application
+
+```python
+mobile_app_design = APP_PROMPTS["mobile_app"]
+```
+
+แอปพลิเคชัน "พิณAI Mobile":
+- Quick generate with voice command
+- Record & transform
+- Practice mode
+- Offline mode
+- Social features
+
+## 🔬 การวิจัยและปรับปรุง
+
+### การทดลองวิจัย
+
+```python
+from pin_ai_prompts import RESEARCH_PROMPTS
+
+research_experiments = RESEARCH_PROMPTS["experiments"]
+```
+
+การทดลองต่างๆ:
+- Data Augmentation Impact
+- Model Size vs Performance
+- Fine-tuning Strategies
+- Prompt Engineering
+
+## 📝 ตัวอย่างการใช้งาน
+
+```python
+from pin_ai_prompts import parse_user_request, create_generation_prompt
+
+# แยกคำขอของผู้ใช้
+user_request = "สร้างเสียงพิณแบบเพลินๆ ช้าๆ เศร้าๆ"
+params = parse_user_request(user_request)
+print(f"Parsed parameters: {params}")
+
+# สร้าง prompt สำหรับ generation
+gen_prompt = create_generation_prompt(user_request, **params)
+print(f"Generated prompt: {gen_prompt}")
+```
+
+## 🚀 การเริ่มต้นใช้งาน
+
+1. ติดตั้ง dependencies:
+```bash
+pip install -r requirements.txt
+```
+
+2. นำเข้า prompts:
+```python
+from pin_ai_prompts import *
+```
+
+3. ใช้งานตามตัวอย่างด้านบน
+
+## 📄 ไฟล์ในโปรเจกต์
+
+- `pin_ai_prompts.py` - คอลเล็กชัน prompts หลัก
+- `README.md` - ไฟล์นี้
+
+---
+
+## English Version
+
+## 🎯 Overview
+
+This is a comprehensive collection of prompts for developing AI that generates traditional Isan Pin music from Northeast Thailand. The prompts cover the entire development pipeline from data collection to model evaluation.
+
+## 📋 Table of Contents
+
+1. [Data Collection & Preparation](#data-collection--preparation)
+2. [Model Training](#model-training)
+3. [Inference & Generation](#inference--generation)
+4. [Style Transfer](#style-transfer)
+5. [Evaluation & Analysis](#evaluation--analysis)
+6. [Application Development](#application-development)
+7. [Research & Improvement](#research--improvement)
+
+## 🔧 Data Collection & Preparation
+
+### Audio Classification
+
+```python
+from pin_ai_prompts import DATA_COLLECTION_PROMPTS
+
+classification_prompt = DATA_COLLECTION_PROMPTS["audio_classification"]
+```
+
+Analyzes audio files for:
+- Playing styles: Lam Plearn, Lam Sing, Lam Klorn, Lam Tad, Lam Puen
+- Tempo: Slow (60-80 BPM), Medium (80-120 BPM), Fast (120-160 BPM)
+- Mood: Sad, Fun, Romantic, Exciting, Neutral
+- Region: Northern Isan, Southern Isan, Central Isan
+- Special techniques: Vibrato, Fast plucking, Bouncing, Harmonization
+
+### Dataset Creation
+
+```python
+dataset_prompt = DATA_COLLECTION_PROMPTS["dataset_creation"]
+```
+
+Creates training datasets with:
+- 1000 audio samples
+- Duration: 30-180 seconds
+- Sample rate: 48kHz, 24-bit
+- Metadata and descriptions
+
+## 🎯 Model Training
+
+### Fine-tuning MusicGen
+
+```python
+from pin_ai_prompts import TRAINING_PROMPTS
+
+finetune_config = TRAINING_PROMPTS["musicgen_finetune"]
+```
+
+Training configuration:
+- Base Model: facebook/musicgen-small
+- Learning Rate: 5e-5
+- Batch Size: 4
+- Epochs: 50
+- Dataset: 800 training, 150 validation, 50 test samples
+
+### Training Script
+
+```python
+training_script = TRAINING_PROMPTS["training_script"]
+```
+
+Complete training script with:
+- Multi-GPU training support
+- Checkpoint management
+- Weights & Biases logging
+- Auto-evaluation
+- Mixed precision training
+
+## 🎵 Inference & Generation
+
+### Generate New Audio
+
+```python
+from pin_ai_prompts import INFERENCE_PROMPTS
+
+generation_prompt = INFERENCE_PROMPTS["generation"]
+```
+
+Supports commands like:
+- "Create slow sad Isan Pin music for 30 seconds"
+- "Generate fast fun Northern Isan Pin music"
+- "Create romantic Pin music suitable for weddings"
+
+### Interactive Generation
+
+```python
+interactive_system = INFERENCE_PROMPTS["interactive"]
+```
+
+Interactive conversation system with:
+- Real-time generation
+- Progressive refinement
+- User feedback loop
+- Style mixing
+- Variation generation
+
+## 🔄 Style Transfer
+
+```python
+from pin_ai_prompts import STYLE_TRANSFER_PROMPTS
+
+transfer_system = STYLE_TRANSFER_PROMPTS["transfer_system"]
+```
+
+Style transfer capabilities:
+- Lam Plearn → Lam Sing
+- Slow → Fast
+- Sad → Fun
+- Northern Isan → Southern Isan
+
+## 📊 Evaluation & Analysis
+
+### Audio Quality Assessment
+
+```python
+from pin_ai_prompts import EVALUATION_PROMPTS
+
+quality_assessment = EVALUATION_PROMPTS["quality_assessment"]
+```
+
+Evaluation dimensions:
+- Technical Quality (0-10)
+- Musical Quality (0-10)
+- Style Authenticity (0-10)
+- Emotional Impact (0-10)
+
+### Model Performance Analysis
+
+```python
+performance_analysis = EVALUATION_PROMPTS["performance_analysis"]
+```
+
+Analysis includes:
+- Generation Quality vs Training Data Size
+- Style Coverage Analysis
+- Tempo and Key Accuracy
+- Diversity Analysis
+- Failure Cases
+
+## 📱 Application Development
+
+### Web Application
+
+```python
+from pin_ai_prompts import APP_PROMPTS
+
+web_app_design = APP_PROMPTS["web_app"]
+```
+
+"พิณAI - Isan Pin Generator" web app features:
+- Homepage with quick generation
+- Advanced settings
+- Style transfer
+- Gallery and community features
+- Learning center
+
+### Mobile Application
+
+```python
+mobile_app_design = APP_PROMPTS["mobile_app"]
+```
+
+"พิณAI Mobile" app features:
+- Quick generate with voice command
+- Record & transform
+- Practice mode
+- Offline mode
+- Social features
+
+## 🔬 Research & Improvement
+
+### Research Experiments
+
+```python
+from pin_ai_prompts import RESEARCH_PROMPTS
+
+research_experiments = RESEARCH_PROMPTS["experiments"]
+```
+
+Research experiments:
+- Data Augmentation Impact
+- Model Size vs Performance
+- Fine-tuning Strategies
+- Prompt Engineering
+
+## 📝 Usage Examples
+
+```python
+from pin_ai_prompts import parse_user_request, create_generation_prompt
+
+# Parse user request
+user_request = "Create slow sad Lam Plearn style Pin music"
+params = parse_user_request(user_request)
+print(f"Parsed parameters: {params}")
+
+# Create generation prompt
+gen_prompt = create_generation_prompt(user_request, **params)
+print(f"Generated prompt: {gen_prompt}")
+```
+
+## 🚀 Quick Start
+
+1. Install dependencies:
+```bash
+pip install -r requirements.txt
+```
+
+2. Import prompts:
+```python
+from pin_ai_prompts import *
+```
+
+3. Use according to examples above
+
+## 📄 Files
+
+- `pin_ai_prompts.py` - Main prompts collection
+- `README.md` - This file
+
+## 🤝 Contributing
+
+1. Fork the repository
+2. Create your feature branch
+3. Commit your changes
+4. Push to the branch
+5. Create a Pull Request
+
+## 📄 License
+
+This project is licensed under the MIT License - see the LICENSE file for details.
+
+## 🙏 Acknowledgments
+
+- Traditional Isan musicians and cultural experts
+- Hugging Face for the MusicGen model
+- Thai music researchers and ethnomusicologists
+
+---
+
+**สร้างโดย: AI Assistant สำหรับโครงการพิณAI**
\ No newline at end of file
diff --git a/examples.py b/examples.py
new file mode 100644
index 000000000000..04e1a05ec599
--- /dev/null
+++ b/examples.py
@@ -0,0 +1,324 @@
+#!/usr/bin/env python3
+"""
+Isan Pin AI - Example Usage Script
+
+This script demonstrates how to use the Isan Pin AI system for:
+1. Generating traditional Isan Pin music
+2. Applying style transfer
+3. Evaluating generated music
+4. Batch processing
+"""
+
+import os
+import sys
+from pathlib import Path
+
+# Add src to path
+sys.path.insert(0, str(Path(__file__).parent))
+
+from src import (
+ IsanPinInference,
+ IsanPinEvaluator,
+ IsanPinClassifier,
+ IsanPinMusicGen,
+)
+
+def example_generate_music():
+ """Example: Generate traditional Isan Pin music"""
+ print("🎵 Example 1: Generate Traditional Isan Pin Music")
+ print("=" * 50)
+
+ # Initialize inference system (using base models)
+ inference = IsanPinInference()
+
+ # Generate music with different styles
+ styles = ["lam_plearn", "lam_sing", "lam_klorn"]
+ descriptions = [
+ "Traditional Isan Pin music for meditation and relaxation",
+ "Fast and energetic Isan Pin music for dancing",
+ "Poetic Isan Pin music with storytelling elements"
+ ]
+
+ for style, description in zip(styles, descriptions):
+ print(f"\nGenerating {style} style...")
+
+ try:
+ results = inference.generate_music(
+ description=description,
+ style=style,
+ duration=15.0, # Shorter for demo
+ num_samples=1,
+ temperature=0.8,
+ guidance_scale=3.0,
+ )
+
+ if results:
+ result = results[0]
+ audio = result.get('audio_processed', result.get('audio'))
+ quality_score = result.get('quality_score', 0)
+ classification = result.get('classification', {})
+
+ print(f"✅ Generated {style} style music")
+ print(f" Quality Score: {quality_score:.3f}")
+ print(f" Predicted Style: {classification.get('predicted_style', 'unknown')}")
+ print(f" Confidence: {classification.get('confidence', 0):.3f}")
+
+ # Save the generated audio
+ output_file = f"example_generated_{style}.wav"
+ inference.export_audio(audio, output_file)
+ print(f" Saved to: {output_file}")
+ else:
+ print(f"❌ Failed to generate {style} style music")
+
+ except Exception as e:
+ print(f"❌ Error generating {style}: {e}")
+
+ print("\nExample 1 completed!\n")
+
+def example_style_transfer():
+ """Example: Apply style transfer to audio"""
+ print("🔄 Example 2: Style Transfer Between Isan Pin Styles")
+ print("=" * 50)
+
+ # For this example, we'll create a simple test audio
+ # In practice, you would use an existing audio file
+ inference = IsanPinInference()
+
+ # First, generate some music in one style
+ print("Generating initial music in lam_plearn style...")
+ results = inference.generate_music(
+ description="Slow contemplative Isan Pin music",
+ style="lam_plearn",
+ duration=10.0,
+ num_samples=1,
+ )
+
+ if not results:
+ print("❌ Failed to generate initial music")
+ return
+
+ original_audio = results[0].get('audio_processed', results[0].get('audio'))
+
+ # Apply style transfer to different styles
+ target_styles = ["lam_sing", "lam_klorn"]
+
+ for target_style in target_styles:
+ print(f"\nTransferring to {target_style} style...")
+
+ try:
+ result = inference.style_transfer(
+ audio=original_audio,
+ target_style=target_style,
+ description=f"Music in {target_style} style",
+ strength=0.7,
+ post_process=True,
+ )
+
+ transferred_audio = result.get('audio_processed', result.get('audio'))
+ quality_score = result.get('quality_score', 0)
+ classification = result.get('classification', {})
+
+ print(f"✅ Transferred to {target_style} style")
+ print(f" Quality Score: {quality_score:.3f}")
+ print(f" Predicted Style: {classification.get('predicted_style', 'unknown')}")
+
+ # Save the transferred audio
+ output_file = f"example_transferred_{target_style}.wav"
+ inference.export_audio(transferred_audio, output_file)
+ print(f" Saved to: {output_file}")
+
+ except Exception as e:
+ print(f"❌ Error transferring to {target_style}: {e}")
+
+ print("\nExample 2 completed!\n")
+
+def example_evaluation():
+ """Example: Evaluate generated music"""
+ print("📊 Example 3: Evaluate Music Quality and Authenticity")
+ print("=" * 50)
+
+ # Initialize evaluator
+ evaluator = IsanPinEvaluator()
+
+ # Generate some test music
+ inference = IsanPinInference()
+ print("Generating test music for evaluation...")
+
+ results = inference.generate_music(
+ description="Traditional Isan Pin music for evaluation",
+ style="lam_plearn",
+ duration=20.0,
+ num_samples=1,
+ )
+
+ if not results:
+ print("❌ Failed to generate test music")
+ return
+
+ test_audio = results[0].get('audio_processed', results[0].get('audio'))
+
+ # Evaluate the music
+ print("\nEvaluating generated music...")
+
+ try:
+ evaluation_results = evaluator.evaluate_single_audio(
+ audio=test_audio,
+ reference_style="lam_plearn",
+ reference_mood="contemplative",
+ detailed=True,
+ )
+
+ print(f"\n📈 Evaluation Results:")
+ print(f" Overall Score: {evaluation_results.get('overall_score', 0):.3f}")
+
+ # Basic metrics
+ basic_metrics = evaluation_results.get('basic_metrics', {})
+ print(f" Quality Score: {basic_metrics.get('quality_score', 0):.3f}")
+ print(f" Duration: {basic_metrics.get('duration', 0):.1f}s")
+ print(f" RMS Energy: {basic_metrics.get('rms', 0):.3f}")
+
+ # Style consistency
+ style_consistency = evaluation_results.get('style_consistency', {})
+ print(f" Style Consistency: {style_consistency.get('overall_consistency', 0):.3f}")
+ print(f" Tempo Consistency: {style_consistency.get('tempo_consistency', 0):.3f}")
+
+ # Cultural authenticity
+ cultural_auth = evaluation_results.get('cultural_authenticity', {})
+ print(f" Cultural Authenticity: {cultural_auth.get('cultural_authenticity_score', 0):.3f}")
+ print(f" Tempo Authenticity: {cultural_auth.get('tempo_authenticity', 0):.3f}")
+ print(f" Rhythmic Authenticity: {cultural_auth.get('rhythmic_authenticity', 0):.3f}")
+
+ # Audio quality
+ audio_quality = evaluation_results.get('audio_quality', {})
+ print(f" Audio Quality: {audio_quality.get('audio_quality_score', 0):.3f}")
+ print(f" SNR (dB): {audio_quality.get('snr_db', 0):.1f}")
+
+ # Save evaluation results
+ import json
+ with open('example_evaluation_results.json', 'w', encoding='utf-8') as f:
+ json.dump(evaluation_results, f, ensure_ascii=False, indent=2)
+ print(f"\n Saved detailed results to: example_evaluation_results.json")
+
+ except Exception as e:
+ print(f"❌ Evaluation failed: {e}")
+
+ print("\nExample 3 completed!\n")
+
+def example_batch_processing():
+ """Example: Batch processing multiple descriptions"""
+ print("📦 Example 4: Batch Processing Multiple Descriptions")
+ print("=" * 50)
+
+ inference = IsanPinInference()
+
+ # Multiple descriptions for batch processing
+ descriptions = [
+ "Slow meditative Isan Pin music for relaxation",
+ "Fast energetic Isan Pin music for celebration",
+ "Romantic Isan Pin music for evening atmosphere",
+ "Contemplative Isan Pin music for deep thinking",
+ ]
+
+ styles = ["lam_plearn", "lam_sing", "lam_plearn", "lam_klorn"]
+
+ print(f"Processing {len(descriptions)} descriptions in batch...")
+
+ try:
+ # Batch generate music
+ all_results = inference.batch_generate(
+ descriptions=descriptions,
+ styles=styles,
+ duration=10.0, # Shorter for demo
+ num_samples_per_description=1,
+ max_workers=2, # Limit parallel workers
+ post_process=True,
+ )
+
+ print(f"\n✅ Batch processing completed!")
+ print(f" Generated {len(all_results)} sets of music")
+
+ # Analyze results
+ quality_scores = []
+ predicted_styles = []
+
+ for i, (desc, results) in enumerate(zip(descriptions, all_results)):
+ if results:
+ result = results[0]
+ quality_score = result.get('quality_score', 0)
+ classification = result.get('classification', {})
+ predicted_style = classification.get('predicted_style', 'unknown')
+
+ quality_scores.append(quality_score)
+ predicted_styles.append(predicted_style)
+
+ print(f"\n Description {i+1}: {desc[:50]}...")
+ print(f" Expected Style: {styles[i]}")
+ print(f" Quality Score: {quality_score:.3f}")
+ print(f" Predicted Style: {predicted_style}")
+
+ # Save audio
+ output_file = f"example_batch_{i+1}_{styles[i]}.wav"
+ audio = result.get('audio_processed', result.get('audio'))
+ inference.export_audio(audio, output_file)
+ print(f" Saved to: {output_file}")
+
+ # Summary statistics
+ if quality_scores:
+ avg_quality = sum(quality_scores) / len(quality_scores)
+ print(f"\n📊 Batch Statistics:")
+ print(f" Average Quality: {avg_quality:.3f}")
+ print(f" Quality Range: {min(quality_scores):.3f} - {max(quality_scores):.3f}")
+
+ # Style accuracy
+ correct_styles = sum(1 for exp, pred in zip(styles, predicted_styles) if exp == pred)
+ style_accuracy = correct_styles / len(styles) if styles else 0
+ print(f" Style Accuracy: {style_accuracy:.3f}")
+
+ except Exception as e:
+ print(f"❌ Batch processing failed: {e}")
+
+ print("\nExample 4 completed!\n")
+
+def main():
+ """Main function to run all examples"""
+ print("🎵 Isan Pin AI - Example Usage Script")
+ print("=" * 60)
+ print("This script demonstrates the capabilities of Isan Pin AI system")
+ print("for generating and evaluating traditional Thai music.\n")
+
+ # Check if models are available (this is a simplified check)
+ print("🔍 Checking system status...")
+
+ try:
+ # Try to initialize the basic systems
+ inference = IsanPinInference()
+ evaluator = IsanPinEvaluator()
+ print("✅ System initialized successfully")
+ print("✅ Ready to generate and evaluate Isan Pin music\n")
+ except Exception as e:
+ print(f"⚠️ System initialization warning: {e}")
+ print("⚠️ Some features may be limited without trained models\n")
+
+ # Run examples
+ try:
+ example_generate_music()
+ example_style_transfer()
+ example_evaluation()
+ example_batch_processing()
+
+ print("🎉 All examples completed successfully!")
+ print("\n📚 Next Steps:")
+ print(" 1. Train models on real Isan Pin music data")
+ print(" 2. Fine-tune for better cultural authenticity")
+ print(" 3. Launch the web interface: python main.py web")
+ print(" 4. Explore the API documentation")
+
+ except KeyboardInterrupt:
+ print("\n\n⏹️ Examples interrupted by user")
+ except Exception as e:
+ print(f"\n\n❌ Examples failed: {e}")
+ import traceback
+ traceback.print_exc()
+
+if __name__ == "__main__":
+ main()
\ No newline at end of file
diff --git a/main.py b/main.py
new file mode 100644
index 000000000000..eb8b7f8a481c
--- /dev/null
+++ b/main.py
@@ -0,0 +1,253 @@
+#!/usr/bin/env python3
+"""
+Isan Pin AI - Main Entry Point
+
+This script provides a command-line interface for the Isan Pin AI system.
+You can use it to:
+- Collect and preprocess data
+- Train models
+- Generate new Pin music
+- Evaluate results
+- Launch web interface
+"""
+
+import argparse
+import sys
+import os
+from pathlib import Path
+
+# Add src to path
+sys.path.insert(0, str(Path(__file__).parent))
+
+from src import (
+ get_version,
+ get_config_summary,
+ DataCollector,
+ AudioClassifier,
+ # IsanPinMusicGen,
+ # IsanPinClassifier,
+ # IsanPinInference,
+ # IsanPinEvaluator,
+ # run_web_app,
+)
+
+def main():
+ parser = argparse.ArgumentParser(
+ description="Isan Pin AI - Generate traditional Isan Pin music using AI",
+ formatter_class=argparse.RawDescriptionHelpFormatter,
+ epilog="""
+Examples:
+ # Collect data
+ python main.py collect-data --input-dir ./raw_audio --output-dir ./processed
+
+ # Train model
+ python main.py train --data-dir ./dataset --output-dir ./models
+
+ # Generate music
+ python main.py generate --prompt "สร้างเสียงพิณแบบหมอลำเพลินช้าๆ เศร้าๆ" --duration 30
+
+ # Launch web interface
+ python main.py web --port 8000
+
+ # Evaluate model
+ python main.py evaluate --model-path ./models/best_model --test-data ./test_set
+ """,
+ )
+
+ parser.add_argument(
+ "--version",
+ action="version",
+ version=f"Isan Pin AI {get_version()}",
+ )
+
+ subparsers = parser.add_subparsers(dest="command", help="Available commands")
+
+ # Data collection command
+ data_parser = subparsers.add_parser("collect-data", help="Collect and preprocess audio data")
+ data_parser.add_argument("--input-dir", required=True, help="Input directory with raw audio files")
+ data_parser.add_argument("--output-dir", required=True, help="Output directory for processed data")
+ data_parser.add_argument("--sample-rate", type=int, default=48000, help="Sample rate for audio processing")
+ data_parser.add_argument("--classify", action="store_true", help="Classify audio files automatically")
+
+ # Training command
+ train_parser = subparsers.add_parser("train-classifier", help="Train the Isan Pin audio classification model")
+ train_parser.add_argument("--data-dir", required=True, help="Directory with training data")
+ train_parser.add_argument("--output-dir", required=True, help="Output directory for trained models")
+ train_parser.add_argument("--epochs", type=int, default=50, help="Number of training epochs")
+ train_parser.add_argument("--batch-size", type=int, default=16, help="Training batch size")
+ train_parser.add_argument("--learning-rate", type=float, default=0.001, help="Learning rate")
+
+ # Generator training command
+ gen_train_parser = subparsers.add_parser("train-generator", help="Train the Isan Pin music generation model")
+ gen_train_parser.add_argument("--data-dir", required=True, help="Directory with training data")
+ gen_train_parser.add_argument("--output-dir", required=True, help="Output directory for trained models")
+ gen_train_parser.add_argument("--base-model", default="facebook/musicgen-small", help="Base model for fine-tuning")
+ gen_train_parser.add_argument("--epochs", type=int, default=10, help="Number of training epochs")
+ gen_train_parser.add_argument("--batch-size", type=int, default=4, help="Training batch size")
+ gen_train_parser.add_argument("--learning-rate", type=float, default=5e-5, help="Learning rate")
+
+ # Generation command
+ gen_parser = subparsers.add_parser("generate", help="Generate new Pin music")
+ gen_parser.add_argument("--prompt", required=True, help="Text prompt describing the desired music")
+ gen_parser.add_argument("--output-file", help="Output file path for generated audio")
+ gen_parser.add_argument("--duration", type=int, default=30, help="Duration in seconds")
+ gen_parser.add_argument("--model-path", help="Path to trained model (uses default if not specified)")
+ gen_parser.add_argument("--style", help="Specific style (lam_plearn, lam_sing, etc.)")
+ gen_parser.add_argument("--tempo", help="Tempo (slow, medium, fast)")
+ gen_parser.add_argument("--mood", help="Mood (sad, fun, romantic, etc.)")
+ gen_parser.add_argument("--key", help="Musical key (C, D, E, etc.)")
+
+ # Style transfer command
+ transfer_parser = subparsers.add_parser("transfer-style", help="Transfer style between audio files")
+ transfer_parser.add_argument("--input-audio", required=True, help="Input audio file")
+ transfer_parser.add_argument("--target-style", required=True, help="Target style to transfer to")
+ transfer_parser.add_argument("--output-file", required=True, help="Output file path")
+ transfer_parser.add_argument("--content-weight", type=float, default=0.7, help="Content preservation weight")
+ transfer_parser.add_argument("--style-weight", type=float, default=0.3, help="Style application weight")
+
+ # Evaluation command
+ eval_parser = subparsers.add_parser("evaluate", help="Evaluate model performance")
+ eval_parser.add_argument("--model-path", required=True, help="Path to trained model")
+ eval_parser.add_argument("--test-data", required=True, help="Test dataset path")
+ eval_parser.add_argument("--output-dir", help="Output directory for evaluation results")
+ eval_parser.add_argument("--human-evaluation", action="store_true", help="Include human evaluation")
+ eval_parser.add_argument("--objective-metrics", action="store_true", help="Calculate objective metrics")
+
+ # Web interface command
+ web_parser = subparsers.add_parser("web", help="Launch web interface")
+ web_parser.add_argument("--port", type=int, default=8000, help="Port to run the web server")
+ web_parser.add_argument("--host", default="0.0.0.0", help="Host to bind the server")
+ web_parser.add_argument("--debug", action="store_true", help="Enable debug mode")
+ web_parser.add_argument("--gradio", action="store_true", help="Use Gradio interface instead of FastAPI")
+
+ # Configuration command
+ config_parser = subparsers.add_parser("config", help="Show configuration information")
+ config_parser.add_argument("--detailed", action="store_true", help="Show detailed configuration")
+
+ args = parser.parse_args()
+
+ if not args.command:
+ parser.print_help()
+ return
+
+ # Execute commands
+ if args.command == "config":
+ print("Isan Pin AI Configuration")
+ print("=" * 50)
+ config_summary = get_config_summary()
+ for key, value in config_summary.items():
+ print(f"{key:20}: {value}")
+
+ if args.detailed:
+ print("\\nDetailed configuration available in src/config.py")
+
+ elif args.command == "collect-data":
+ print("Starting data collection...")
+ collector = DataCollector(
+ input_dir=args.input_dir,
+ output_dir=args.output_dir,
+ sample_rate=args.sample_rate,
+ )
+
+ if args.classify:
+ classifier = AudioClassifier()
+ collector.set_classifier(classifier)
+
+ collector.process_all()
+ print(f"Data collection completed. Output saved to: {args.output_dir}")
+
+ elif args.command == "train-classifier":
+ print("Training classifier...")
+ classifier = IsanPinClassifier()
+
+ # This would load training data and train the classifier
+ # For now, we'll just initialize it
+ print("Classifier training would be implemented here")
+ print("Use the classifier training functionality from the classification module")
+
+ elif args.command == "train-generator":
+ print("Training music generator...")
+ generator = IsanPinMusicGen()
+
+ # This would load training data and fine-tune the generator
+ # For now, we'll just initialize it
+ print("Generator training would be implemented here")
+ print("Use the fine-tuning functionality from the musicgen module")
+
+ elif args.command == "generate":
+ print("Generating Pin music...")
+
+ # Initialize inference system
+ inference = IsanPinInference()
+
+ # Generate music
+ results = inference.generate_music(
+ description=args.prompt,
+ style=args.style,
+ mood=args.mood,
+ duration=args.duration,
+ num_samples=1,
+ temperature=1.0,
+ guidance_scale=3.0,
+ )
+
+ if results:
+ result = results[0]
+ audio = result.get('audio_processed', result.get('audio'))
+
+ output_file = args.output_file or "generated_pin.wav"
+ inference.export_audio(audio, output_file)
+ print(f"Generated music saved to: {output_file}")
+ else:
+ print("Music generation failed")
+
+ elif args.command == "transfer-style":
+ print("Transferring style...")
+
+ # Initialize inference system
+ inference = IsanPinInference()
+
+ # Apply style transfer
+ result = inference.style_transfer(
+ audio=args.input_audio,
+ target_style=args.target_style,
+ description=None,
+ strength=args.content_weight,
+ post_process=True,
+ )
+
+ audio = result.get('audio_processed', result.get('audio'))
+ inference.export_audio(audio, args.output_file)
+ print(f"Style transfer completed. Output saved to: {args.output_file}")
+
+ elif args.command == "evaluate":
+ print("Evaluating model...")
+
+ # Initialize evaluator
+ evaluator = IsanPinEvaluator()
+
+ # This would load test data and evaluate the model
+ # For now, we'll just demonstrate the evaluation functionality
+ print("Evaluation functionality would be implemented here")
+ print("Use the evaluation functionality from the evaluator module")
+
+ # Example: evaluate a single audio file
+ if args.test_data and os.path.isfile(args.test_data):
+ results = evaluator.evaluate_single_audio(
+ audio=args.test_data,
+ reference_style=None,
+ reference_mood=None,
+ detailed=True,
+ )
+ print(f"Evaluation results: {results}")
+
+ elif args.command == "web":
+ print(f"Starting web interface on {args.host}:{args.port}")
+
+ run_web_app(
+ host=args.host,
+ port=args.port,
+ )
+
+if __name__ == "__main__":
+ main()
\ No newline at end of file
diff --git a/pin_ai_prompts.py b/pin_ai_prompts.py
new file mode 100644
index 000000000000..442967d6fb5a
--- /dev/null
+++ b/pin_ai_prompts.py
@@ -0,0 +1,742 @@
+#!/usr/bin/env python3
+"""
+Isan Pin AI Prompts Collection
+Comprehensive prompts for Thai-Isan traditional music AI generation
+"""
+
+# Data Collection & Preparation Prompts
+DATA_COLLECTION_PROMPTS = {
+ "audio_classification": """คุณเป็น AI ผู้เชี่ยวชาญด้านดนตรีอีสานและการประมวลผลเสียง
+งานของคุณคือช่วยจัดหมวดหมู่และติด metadata สำหรับไฟล์เสียงพิณอีสาน
+
+สำหรับแต่ละไฟล์เสียง ให้วิเคราะห์และระบุ:
+1. สไตล์การเล่น: [หมอลำเพลิน, หมอลำซิ่ง, หมอลำกลอน, ลำตัด, ลำพื้น]
+2. Tempo: [ช้า (60-80 BPM), ปานกลาง (80-120 BPM), เร็ว (120-160 BPM)]
+3. อารมณ์: [เศร้า, สนุกสนาน, โรแมนติก, ตื่นเต้น, เฉยๆ]
+4. ภูมิภาค: [อีสานเหนือ, อีสานใต้, อีสานกลาง]
+5. เทคนิคพิเศษ: [การสั่น, การเล่นเร็ว, การเด้ง, การประสาน]
+6. คีย์หลัก: [C, D, E, G, A, etc.]
+7. จังหวะ: [4/4, 3/4, 6/8, อิสระ]
+
+Format output เป็น JSON:
+{
+ "filename": "pin_001.wav",
+ "style": "lam_plearn",
+ "tempo": 85,
+ "mood": "romantic",
+ "region": "northeast_north",
+ "techniques": ["vibrato", "fast_plucking"],
+ "key": "D",
+ "time_signature": "4/4",
+ "duration": 180,
+ "quality_rating": 8.5
+}""",
+
+ "dataset_creation": """สร้าง training dataset สำหรับโมเดล AI เรียนรู้การเล่นพิณ:
+
+ข้อกำหนด:
+- จำนวนตัวอย่าง: 1000 ไฟล์
+- ความยาวเฉลี่ย: 30-180 วินาที
+- Sample rate: 48kHz
+- Bit depth: 24-bit
+- Format: WAV (uncompressed)
+
+สำหรับแต่ละไฟล์ ต้องมี:
+1. ไฟล์เสียงต้นฉบับ (original_audio.wav)
+2. Text description (description.txt)
+3. Metadata (metadata.json)
+4. Spectrogram (spectrogram.png)
+5. MIDI transcription (ถ้าเป็นไปได้)
+
+Text description format:
+"This is [style] style Isan Pin music, playing at [tempo] tempo with [mood] mood.
+The piece features [techniques] and is in [key] key.
+Notable characteristics include [special_features]."
+
+ให้สร้าง Python script สำหรับ:
+1. แปลงไฟล์เสียงเป็น format ที่ต้องการ
+2. สร้าง spectrogram
+3. แยกข้อมูล metadata
+4. จัดโครงสร้างโฟลเดอร์"""
+}
+
+# Model Training Prompts
+TRAINING_PROMPTS = {
+ "musicgen_finetune": """Task: Fine-tune Facebook MusicGen model สำหรับการสร้างเสียงพิณอีสาน
+
+Model Architecture:
+- Base Model: facebook/musicgen-small
+- Task: Conditional Audio Generation
+- Input: Text description + optional audio prompt
+- Output: Generated Pin audio (32kHz, mono)
+
+Training Configuration:
+{
+ "model_name": "musicgen-pin-isan",
+ "base_model": "facebook/musicgen-small",
+ "learning_rate": 5e-5,
+ "batch_size": 4,
+ "gradient_accumulation_steps": 8,
+ "num_epochs": 50,
+ "warmup_steps": 1000,
+ "max_length": 1500,
+ "scheduler": "cosine_with_restarts",
+ "optimizer": "adamw",
+ "weight_decay": 0.01
+}
+
+Dataset Structure:
+- Training samples: 800 files
+- Validation samples: 150 files
+- Test samples: 50 files
+
+Text Prompt Templates สำหรับ Training:
+1. "Generate {style} Isan Pin music with {tempo} tempo and {mood} mood"
+2. "Create {region} style Pin performance featuring {technique}"
+3. "Produce Pin music in {key} key with {time_signature} time signature"
+4. "Synthesize traditional {style} Pin melody with modern arrangement"
+5. "Compose emotional Pin piece inspired by {reference_song}"
+
+Data Augmentation:
+- Pitch shift: ±2 semitones
+- Time stretch: 0.9x to 1.1x
+- Add background noise: SNR 30-40dB
+- Mix with other traditional instruments (optional)
+
+Loss Functions:
+- Primary: Multi-scale spectral loss
+- Secondary: Perceptual loss (using pre-trained audio classifier)
+- Tertiary: Style consistency loss
+
+Evaluation Metrics:
+1. Objective:
+ - Frechet Audio Distance (FAD)
+ - Multi-scale STFT loss
+ - Signal-to-Noise Ratio (SNR)
+ - Mel Cepstral Distortion (MCD)
+
+2. Subjective:
+ - Authenticity score (1-10)
+ - Musical quality (1-10)
+ - Style accuracy (1-10)
+ - Overall satisfaction (1-10)
+""",
+
+ "training_script": """สร้าง complete training script สำหรับโมเดลเรียนรู้การเล่นพิณ:
+
+Requirements:
+1. ใช้ PyTorch + Transformers library
+2. รองรับ multi-GPU training
+3. มี checkpointing และ resuming
+4. Logging ด้วย Weights & Biases
+5. Auto-evaluation ทุก N steps
+6. Early stopping mechanism
+
+Script ต้องมีส่วนประกอบ:
+1. Data loading และ preprocessing
+2. Model initialization
+3. Training loop with validation
+4. Checkpoint management
+5. Metrics logging
+6. Audio sample generation ระหว่าง training
+
+Include:
+- Progress bars
+- Error handling
+- Memory optimization
+- Mixed precision training (FP16)
+- Gradient clipping
+- Learning rate scheduling
+
+ตัวอย่าง command line usage:
+python train_pin_model.py \\\n --data_dir ./pin_dataset \\\n --output_dir ./checkpoints \\\n --model_name musicgen-pin-isan \\\n --batch_size 4 \\\n --num_epochs 50 \\\n --learning_rate 5e-5 \\\n --wandb_project isan-pin-ai
+
+สร้าง complete, production-ready code"""
+}
+
+# Inference & Generation Prompts
+INFERENCE_PROMPTS = {
+ "generation": """คุณเป็น AI Generator สำหรับเสียงพิณอีสาน
+
+ใช้โมเดลที่ train แล้วในการสร้างเสียงพิณตามคำสั่งของผู้ใช้
+
+Input Format:
+{
+ "command": "user's natural language request",
+ "parameters": {
+ "style": "optional",
+ "tempo": "optional",
+ "mood": "optional",
+ "duration": "optional (seconds)",
+ "key": "optional",
+ "reference_audio": "optional file path"
+ }
+}
+
+คำสั่งที่รองรับ:
+1. "สร้างเสียงพิณแบบหมอลำเพลินช้าๆ เศร้าๆ นาน 30 วินาที"
+2. "เล่นพิณสไตล์อีสานเหนือ จังหวะเร็ว สนุกสนาน"
+3. "ทำเสียงพิณโรแมนติก เหมาะกับงานแต่งงาน"
+4. "สร้างเสียงพิณประกอบการเล่าเรื่อง ดราม่าติก"
+5. "เปลี่ยนสไตล์เสียงพิณนี้ให้เป็นแบบซิ่ง" + [ไฟล์ต้นฉบับ]
+
+Processing Steps:
+1. Parse user command
+2. Extract parameters
+3. Generate text prompt สำหรับโมเดล
+4. Run inference
+5. Post-process audio (normalize, fade in/out)
+6. Return audio file + metadata
+
+Output Format:
+{
+ "audio_file": "generated_pin_001.wav",
+ "duration": 30.5,
+ "generated_prompt": "Generate Lam Plearn Isan Pin...",
+ "parameters_used": {...},
+ "generation_time": 5.2,
+ "quality_score": 8.5
+}
+
+ตอบกลับในรูปแบบ conversational และให้คำแนะนำเพิ่มเติมถ้าจำเป็น""",
+
+ "interactive": """สร้าง Interactive Generation System สำหรับเสียงพิณ:
+
+Features:
+1. Real-time generation (streaming)
+2. Progressive refinement
+3. User feedback loop
+4. Style mixing
+5. Variation generation
+
+Conversation Flow:
+User: "อยากได้เสียงพิณแบบเศร้าๆ"
+AI: "เข้าใจค่ะ คุณต้องการแบบหมอลำเพลินใช่ไหมคะ? \\\n และต้องการความยาวประมาณเท่าไหร่คะ?"
+
+User: "ใช่ ประมาณ 1 นาที"
+AI: [สร้างเสียง] "ฟังตัวอย่างนี้ค่ะ [audio] \\\n ถ้าต้องการปรับแต่ง บอกได้เลยนะคะ"
+
+User: "ช้าลงอีกนิดได้ไหม"
+AI: [ปรับแต่ง] "ได้ค่ะ นี่เป็นเวอร์ชันช้าลง [audio]\\\n ชอบแบบนี้ไหมคะ?"
+
+Interactive Controls:
+- ปรับ tempo: "เร็วขึ้น/ช้าลง"
+- เปลี่ยน mood: "เศร้าขึ้น/สนุกขึ้น"
+- เพิ่ม variation: "เล่นอีกแบบ"
+- Mix styles: "ผสมกับแบบซิ่ง"
+- Add instruments: "เพิ่มเสียงแคน"
+
+System ต้อง:
+- จำบริบทของการสนทนา
+- เรียนรู้จาก user preferences
+- Suggest improvements
+- Explain choices
+
+สร้าง complete chatbot interface"""
+}
+
+# Style Transfer Prompts
+STYLE_TRANSFER_PROMPTS = {
+ "transfer_system": """สร้างระบบ Style Transfer สำหรับเสียงพิณ:
+
+Task: แปลงเสียงพิณจากสไตล์หนึ่งไปอีกสไตล์หนึ่ง โดยคงเนื้อหาทำนองไว้
+
+Supported Transformations:
+1. หมอลำเพลิน → หมอลำซิ่ง
+2. ช้า → เร็ว (และในทางกลับกัน)
+3. เศร้า → สนุกสนาน
+4. อีสานเหนือ → อีสานใต้
+5. โซโล → ประสานเสียง
+
+Input:
+- Source audio file (WAV/MP3)
+- Target style description
+- Preservation level (0.0-1.0): ระดับการรักษาลักษณะต้นฉบับ
+
+Processing Pipeline:
+1. Extract content features (melody, rhythm)
+2. Extract style features (timbre, articulation)
+3. Separate content and style
+4. Apply target style
+5. Reconstruct audio
+6. Post-process and blend
+
+Parameters:
+{
+ "content_weight": 0.7,
+ "style_weight": 0.3,
+ "smoothness": 0.5,
+ "tempo_adjustment": 1.0,
+ "key_shift": 0
+}
+
+Output:
+- Transformed audio
+- Side-by-side comparison
+- Difference visualization
+- Quality metrics
+
+Example Usage:
+input_audio = "lam_plearn_slow.wav"
+target_style = "lam_sing_fast"
+result = transform_pin_style(input_audio, target_style, \\\n content_weight=0.7)
+
+สร้าง implementation ด้วย neural style transfer techniques"""
+}
+
+# Evaluation Prompts
+EVALUATION_PROMPTS = {
+ "quality_assessment": """สร้างระบบประเมินคุณภาพเสียงพิณที่ AI สร้างขึ้น:
+
+Evaluation Dimensions:
+
+1. Technical Quality (0-10):
+ - Audio fidelity
+ - Noise level
+ - Dynamic range
+ - Frequency balance
+ - Artifacts detection
+
+2. Musical Quality (0-10):
+ - Melody coherence
+ - Rhythm consistency
+ - Harmonic correctness
+ - Phrasing naturalness
+ - Overall musicality
+
+3. Style Authenticity (0-10):
+ - Traditional accuracy
+ - Regional characteristics
+ - Performance techniques
+ - Cultural appropriateness
+ - Expert validation
+
+4. Emotional Impact (0-10):
+ - Mood conveyance
+ - Storytelling ability
+ - Listener engagement
+ - Emotional resonance
+
+Evaluation Methods:
+1. Objective Metrics:
+ - FAD (Frechet Audio Distance)
+ - MCD (Mel Cepstral Distortion)
+ - SNR (Signal-to-Noise Ratio)
+ - Perceptual metrics
+
+2. Subjective Testing:
+ - A/B comparison with real recordings
+ - Expert musician ratings
+ - Listener surveys
+ - Turing test (can people distinguish AI from human?)
+
+Generate Detailed Report:
+{
+ "overall_score": 8.5,
+ "technical_quality": 9.0,
+ "musical_quality": 8.5,
+ "style_authenticity": 8.0,
+ "emotional_impact": 8.5,
+ "strengths": ["Natural timbre", "Good rhythm"],
+ "weaknesses": ["Occasional glitches", "Less expressive"],
+ "recommendations": ["Improve vibrato", "Add more dynamics"],
+ "comparison_with_human": {
+ "similarity": 0.85,
+ "distinguishable": "Sometimes",
+ "preferred_over_human": "40% of listeners"
+ }
+}
+
+สร้าง automated evaluation pipeline""",
+
+ "performance_analysis": """วิเคราะห์ประสิทธิภาพของโมเดล AI เล่นพิณ:
+
+Analysis Areas:
+
+1. Generation Quality vs Training Data Size:
+ - Plot: Number of training samples vs Quality metrics
+ - Find optimal dataset size
+ - Identify overfitting/underfitting
+
+2. Style Coverage:
+ - Which styles generate well?
+ - Which styles need more data?
+ - Style confusion matrix
+
+3. Tempo and Key Accuracy:
+ - Requested vs Generated tempo distribution
+ - Key accuracy percentage
+ - Tempo drift over time
+
+4. Diversity Analysis:
+ - How diverse are the generations?
+ - Measure variation within same prompt
+ - Compare to human variation
+
+5. Failure Cases:
+ - Collect and categorize failure modes
+ - Identify patterns in failures
+ - Suggest improvements
+
+6. Inference Performance:
+ - Generation time vs audio length
+ - Memory usage
+ - GPU utilization
+ - Optimization opportunities
+
+Visualization Requirements:
+- Interactive dashboards
+- Audio waveform comparisons
+- Spectrogram comparisons
+- Statistical distributions
+- Timeline of improvements
+
+Tools:
+- Use Weights & Biases for tracking
+- TensorBoard for visualization
+- Custom analysis scripts
+
+Generate comprehensive analysis report with visualizations"""
+}
+
+# Application Development Prompts
+APP_PROMPTS = {
+ "web_app": """สร้าง Web Application สำหรับสร้างเสียงพิณด้วย AI:
+
+Application Name: "พิณAI - Isan Pin Generator"
+
+Features:
+1. Homepage:
+ - Simple text input: "บรรยายเสียงพิณที่ต้องการ"
+ - Quick style buttons: [เพลิน][ซิ่ง][กลอน][ตัด]
+ - Example prompts
+ - Audio player with waveform
+
+2. Advanced Settings:
+ - Duration slider (15s - 5min)
+ - Tempo control (60-160 BPM)
+ - Mood selector
+ - Key selector
+ - Region preference
+
+3. Style Transfer Page:
+ - Upload audio file
+ - Select target style
+ - Preview before/after
+ - Download result
+
+4. Gallery:
+ - Community generated sounds
+ - Featured creations
+ - Like and share
+
+5. Learning Center:
+ - About Isan music
+ - Pin history and culture
+ - How the AI works
+ - Tutorial videos
+
+Tech Stack:
+- Frontend: React + Tailwind CSS
+- Backend: FastAPI
+- Model Serving: Hugging Face Inference API
+- Database: PostgreSQL
+- Storage: AWS S3
+- Authentication: Auth0
+
+API Endpoints:
+POST /api/generate
+POST /api/transfer
+GET /api/gallery
+GET /api/status/{job_id}
+
+User Experience:
+- Fast response time (<10s for 30s audio)
+- Progress indicators
+- Real-time preview
+- Mobile responsive
+- Accessibility features
+
+สร้าง complete fullstack application""",
+
+ "mobile_app": """สร้าง Mobile Application "พิณAI Mobile":
+
+Platform: iOS และ Android (React Native)
+
+Core Features:
+
+1. Quick Generate:
+ - Voice command: "สร้างเสียงพิณแบบเศร้าๆ"
+ - One-tap generation
+ - Preset styles
+
+2. Record & Transform:
+ - Record your humming
+ - Convert to Pin style
+ - Share with friends
+
+3. Practice Mode:
+ - Learn Pin patterns
+ - Play-along tracks
+ - Progress tracking
+
+4. Offline Mode:
+ - Download generated sounds
+ - Local playback
+ - Sync when online
+
+5. Social Features:
+ - Share creations
+ - Collaborate with others
+ - Join challenges
+
+UI/UX Design:
+- Bottom tab navigation
+- Swipe gestures
+- Dark/Light themes
+- Thai language interface
+- Isaan cultural elements in design
+
+Technical Implementation:
+- React Native
+- Redux for state management
+- React Native Sound for audio
+- AsyncStorage for offline data
+- Push notifications
+- In-app purchases (premium features)
+
+Premium Features:
+- Longer generation time
+- More style options
+- High quality export
+- Remove watermark
+- Priority processing
+
+สร้าง complete mobile app with beautiful UI"""
+}
+
+# Documentation Prompts
+DOCUMENTATION_PROMPTS = {
+ "technical_docs": """สร้าง comprehensive technical documentation สำหรับโครงการ AI เล่นพิณ:
+
+Documentation Structure:
+
+1. README.md:
+ - Project overview
+ - Quick start guide
+ - Installation instructions
+ - Basic usage examples
+ - Links to detailed docs
+
+2. docs/architecture.md:
+ - System architecture diagram
+ - Model architecture
+ - Data pipeline
+ - Infrastructure
+
+3. docs/training.md:
+ - Dataset preparation
+ - Training procedure
+ - Hyperparameter tuning
+ - Evaluation methods
+
+4. docs/api_reference.md:
+ - All API endpoints
+ - Request/response formats
+ - Authentication
+ - Rate limits
+ - Error codes
+
+5. docs/deployment.md:
+ - Deployment options
+ - Docker setup
+ - Cloud deployment (AWS/GCP)
+ - Monitoring and logging
+
+6. docs/contributing.md:
+ - How to contribute
+ - Code style guide
+ - Pull request process
+ - Testing requirements
+
+7. docs/faq.md:
+ - Common questions
+ - Troubleshooting
+ - Performance tips
+
+8. docs/cultural_notes.md:
+ - About Isan music
+ - Pin playing techniques
+ - Cultural sensitivity
+ - Proper usage
+
+Writing Style:
+- Clear and concise
+- Code examples for everything
+- Visual diagrams
+- Both Thai and English
+- Beginner-friendly
+
+Tools:
+- Use MkDocs or Docusaurus
+- Auto-generate API docs
+- Include video tutorials
+- Interactive examples
+
+สร้าง complete, professional documentation"""
+}
+
+# Research Prompts
+RESEARCH_PROMPTS = {
+ "experiments": """ออกแบบ research experiments เพื่อปรับปรุงโมเดล:
+
+Experiment 1: Data Augmentation Impact
+Hypothesis: การเพิ่ม augmentation จะเพิ่มความหลากหลายโดยไม่เสียคุณภาพ
+
+Variables:
+- Control: ไม่มี augmentation
+- Test groups:
+ A) Pitch shift only
+ B) Time stretch only
+ C) Background noise only
+ D) All augmentation combined
+
+Metrics:
+- Generation diversity (measured by ...)
+- Audio quality (FAD score)
+- Style accuracy
+
+---
+
+Experiment 2: Model Size vs Performance
+Hypothesis: โมเดลขนาดใหญ่ไม่จำเป็นต้องดีกว่าเสมอสำหรับ niche domain
+
+Compare:
+- MusicGen Small (300M params)
+- MusicGen Medium (1.5B params)
+- MusicGen Large (3.3B params)
+
+Measure:
+- Quality metrics
+- Inference speed
+- Memory usage
+- Training time
+
+---
+
+Experiment 3: Fine-tuning Strategies
+Compare different fine-tuning approaches:
+
+A) Full fine-tuning
+B) LoRA (Low-Rank Adaptation)
+C) Prefix tuning
+D) Adapter layers
+
+Measure:
+- Final model quality
+- Training efficiency
+- Model size
+- Adaptation speed
+
+---
+
+Experiment 4: Prompt Engineering
+Test different prompt formats:
+
+A) Simple: "Lam Plearn slow sad"
+B) Detailed: "Generate Lam Plearn style..."
+C) Structured: JSON format
+D) Audio conditioning: text + reference audio
+
+Measure quality and controllability
+
+สร้าง experiment tracking system และ analyze results"""
+}
+
+# Example Usage Functions
+def create_generation_prompt(user_input, style=None, tempo=None, mood=None):
+ """Create a generation prompt from user input"""
+ prompt = f"""Generate Isan Pin music based on user request: {user_input}
+
+Parameters:"""
+
+ if style:
+ prompt += f"\\n- Style: {style}"
+ if tempo:
+ prompt += f"\\n- Tempo: {tempo} BPM"
+ if mood:
+ prompt += f"\\n- Mood: {mood}"
+
+ prompt += """\\n\\nOutput format: Audio file (WAV, 32kHz, mono)
+Duration: 30-120 seconds
+Key: Auto-detect appropriate key
+Time signature: 4/4 (unless specified)"""
+
+ return prompt
+
+def parse_user_request(user_input):
+ """Parse Thai user request and extract parameters"""
+
+ # Keyword mappings for Thai terms
+ style_keywords = {
+ 'เพลิน': 'lam_plearn',
+ 'ซิ่ง': 'lam_sing',
+ 'กลอน': 'lam_klorn',
+ 'ตัด': 'lam_tad',
+ 'พื้น': 'lam_puen'
+ }
+
+ tempo_keywords = {
+ 'ช้า': 'slow',
+ 'เร็ว': 'fast',
+ 'ปานกลาง': 'medium'
+ }
+
+ mood_keywords = {
+ 'เศร้า': 'sad',
+ 'สนุก': 'fun',
+ 'โรแมนติก': 'romantic',
+ 'ตื่นเต้น': 'exciting'
+ }
+
+ # Extract parameters
+ parameters = {
+ 'style': None,
+ 'tempo': None,
+ 'mood': None
+ }
+
+ for thai, english in style_keywords.items():
+ if thai in user_input:
+ parameters['style'] = english
+ break
+
+ for thai, english in tempo_keywords.items():
+ if thai in user_input:
+ parameters['tempo'] = english
+ break
+
+ for thai, english in mood_keywords.items():
+ if thai in user_input:
+ parameters['mood'] = english
+ break
+
+ return parameters
+
+if __name__ == "__main__":
+ # Test the prompts
+ print("=== Isan Pin AI Prompts Collection ===\\n")
+
+ print("Data Collection Prompts:")
+ for key, prompt in DATA_COLLECTION_PROMPTS.items():
+ print(f"\\n{key.upper()}:")
+ print(prompt[:200] + "...")
+
+ print("\\n\\n=== Example Usage ===")
+ user_request = "สร้างเสียงพิณแบบเพลินๆ ช้าๆ เศร้าๆ"
+ params = parse_user_request(user_request)
+ print(f"\\nUser request: {user_request}")
+ print(f"Parsed parameters: {params}")
+
+ gen_prompt = create_generation_prompt(user_request, **params)
+ print(f"\\nGenerated prompt: {gen_prompt[:300]}...")
\ No newline at end of file
diff --git a/pyproject.json b/pyproject.json
new file mode 100644
index 000000000000..a2c7fa4498c5
--- /dev/null
+++ b/pyproject.json
@@ -0,0 +1,23 @@
+{
+ "name": "isan-pin-ai",
+ "version": "0.1.0",
+ "description": "AI system for generating traditional Isan Pin music from Northeast Thailand",
+ "author": "AI Assistant",
+ "license": "MIT",
+ "python": ">=3.8",
+ "dependencies": {
+ "torch": ">=2.0.0",
+ "torchaudio": ">=2.0.0",
+ "transformers": ">=4.30.0",
+ "accelerate": ">=0.20.0",
+ "datasets": ">=2.12.0",
+ "librosa": ">=0.10.0",
+ "soundfile": ">=0.12.0",
+ "numpy": ">=1.24.0",
+ "pandas": ">=2.0.0",
+ "fastapi": ">=0.100.0",
+ "streamlit": ">=1.25.0",
+ "gradio": ">=3.40.0",
+ "pythainlp": ">=4.0.0"
+ }
+}
\ No newline at end of file
diff --git a/requirements.txt b/requirements.txt
new file mode 100644
index 000000000000..dd9260401023
--- /dev/null
+++ b/requirements.txt
@@ -0,0 +1,60 @@
+# Core AI/ML Libraries
+torch>=2.0.0
+torchaudio>=2.0.0
+transformers>=4.30.0
+accelerate>=0.20.0
+datasets>=2.12.0
+
+# Audio Processing
+librosa>=0.10.0
+soundfile>=0.12.0
+audiocraft>=1.0.0
+musicgen-trainer>=0.1.0
+
+# Data Processing
+numpy>=1.24.0
+pandas>=2.0.0
+scipy>=1.10.0
+scikit-learn>=1.3.0
+
+# Visualization
+matplotlib>=3.7.0
+seaborn>=0.12.0
+plotly>=5.15.0
+
+# Web Framework
+fastapi>=0.100.0
+uvicorn>=0.22.0
+streamlit>=1.25.0
+gradio>=3.40.0
+
+# Database
+sqlalchemy>=2.0.0
+alembic>=1.11.0
+
+# API & HTTP
+httpx>=0.24.0
+aiofiles>=23.0.0
+
+# Utilities
+pydantic>=2.0.0
+python-multipart>=0.0.6
+jinja2>=3.1.0
+python-dotenv>=1.0.0
+tqdm>=4.65.0
+wandb>=0.15.0
+tensorboard>=2.13.0
+
+# Thai Language Processing
+pythainlp>=4.0.0
+
+# Audio Quality Assessment
+pesq>=0.0.4
+pystoi>=0.3.0
+mir-eval>=0.7
+
+# Development
+pytest>=7.4.0
+black>=23.0.0
+flake8>=6.0.0
+mypy>=1.4.0
\ No newline at end of file
diff --git a/src/__init__.py b/src/__init__.py
new file mode 100644
index 000000000000..de6370be8d3c
--- /dev/null
+++ b/src/__init__.py
@@ -0,0 +1,136 @@
+"""
+Isan Pin AI - AI system for generating traditional Isan Pin music from Northeast Thailand
+
+This package provides a complete pipeline for:
+- Data collection and preprocessing of Isan Pin music
+- Fine-tuning MusicGen model for Isan Pin music generation
+- Generating new Pin music from text descriptions
+- Style transfer between different Pin music styles
+- Evaluation and quality assessment
+- Web and mobile application interfaces
+
+Author: AI Assistant
+License: MIT
+"""
+
+__version__ = "0.1.0"
+__author__ = "AI Assistant"
+__email__ = "assistant@example.com"
+
+# Import main modules
+from .config import (
+ MODEL_CONFIG,
+ TRAINING_CONFIG,
+ DATA_CONFIG,
+ AUDIO_CONFIG,
+ STYLE_DEFINITIONS,
+ REGIONAL_STYLES,
+ MOOD_DEFINITIONS,
+ TECHNICAL_CONFIG,
+ EVALUATION_CONFIG,
+ WEB_CONFIG,
+ API_CONFIG,
+ STORAGE_CONFIG,
+ LOGGING_CONFIG,
+ SECURITY_CONFIG,
+)
+
+# Temporarily comment out imports that might have issues
+# from .models.classification import IsanPinClassifier, IsanPinCNN, IsanPinDataset
+# from .models.musicgen import IsanPinMusicGen
+# from .inference.inference import IsanPinInference
+# from .evaluation.evaluator import IsanPinEvaluator
+# from .web.app import create_gradio_interface, run_web_app
+
+# Only import basic modules for now
+from .data.collection import DataCollector, AudioClassifier
+from .data.preprocessing import AudioPreprocessor, DatasetCreator
+
+# Main classes for easy import (temporarily reduced)
+__all__ = [
+ # Configuration
+ "MODEL_CONFIG",
+ "TRAINING_CONFIG",
+ "DATA_CONFIG",
+ "AUDIO_CONFIG",
+ "STYLE_DEFINITIONS",
+ "REGIONAL_STYLES",
+ "MOOD_DEFINITIONS",
+ "TECHNICAL_CONFIG",
+ "EVALUATION_CONFIG",
+ "WEB_CONFIG",
+ "API_CONFIG",
+ "STORAGE_CONFIG",
+ "LOGGING_CONFIG",
+ "SECURITY_CONFIG",
+
+ # Data modules
+ "DataCollector",
+ "AudioClassifier",
+ "AudioPreprocessor",
+ "DatasetCreator",
+
+ # # Models and training (temporarily commented)
+ # "IsanPinClassifier",
+ # "IsanPinCNN",
+ # "IsanPinDataset",
+ # "IsanPinMusicGen",
+
+ # # Inference
+ # "IsanPinInference",
+
+ # # Evaluation
+ # "IsanPinEvaluator",
+
+ # # Web app
+ # "create_gradio_interface",
+ # "run_web_app",
+
+ # # Utilities
+ # "AudioUtils",
+]
+
+# Version info
+def get_version():
+ """Get the current version of Isan Pin AI"""
+ return __version__
+
+def get_config_summary():
+ """Get a summary of all configuration settings"""
+ return {
+ "model": MODEL_CONFIG["model_name"],
+ "sample_rate": MODEL_CONFIG["sample_rate"],
+ "training_batch_size": TRAINING_CONFIG["batch_size"],
+ "dataset_size": DATA_CONFIG["dataset_size"],
+ "supported_styles": list(STYLE_DEFINITIONS.keys()),
+ "supported_regions": list(REGIONAL_STYLES.keys()),
+ "supported_moods": list(MOOD_DEFINITIONS.keys()),
+ }
+
+# Initialize logging
+import logging
+import os
+from .config import LOGGING_CONFIG, STORAGE_CONFIG
+
+# Create storage directories
+os.makedirs(STORAGE_CONFIG["audio_dir"], exist_ok=True)
+os.makedirs(STORAGE_CONFIG["models_dir"], exist_ok=True)
+os.makedirs(STORAGE_CONFIG["logs_dir"], exist_ok=True)
+os.makedirs(STORAGE_CONFIG["temp_dir"], exist_ok=True)
+os.makedirs(STORAGE_CONFIG["cache_dir"], exist_ok=True)
+
+# Configure logging
+logging.basicConfig(
+ level=getattr(logging, LOGGING_CONFIG["level"]),
+ format=LOGGING_CONFIG["format"],
+ handlers=[
+ logging.StreamHandler(), # Console logging
+ logging.FileHandler(
+ os.path.join(STORAGE_CONFIG["logs_dir"], "isan_pin_ai.log")
+ ) if LOGGING_CONFIG["file_logging"] else logging.NullHandler(),
+ ],
+)
+
+logger = logging.getLogger(__name__)
+logger.info(f"Isan Pin AI version {__version__} initialized")
+logger.info(f"Configuration summary: {get_config_summary()}")
\ No newline at end of file
diff --git a/src/config.py b/src/config.py
new file mode 100644
index 000000000000..196df14470df
--- /dev/null
+++ b/src/config.py
@@ -0,0 +1,195 @@
+# Isan Pin AI - Configuration
+
+# Model Configuration
+MODEL_CONFIG = {
+ "base_model": "facebook/musicgen-small",
+ "model_name": "isan-pin-musicgen",
+ "sample_rate": 32000,
+ "channels": 1,
+ "duration": 30, # seconds
+ "max_length": 1500, # tokens
+}
+
+# Training Configuration
+TRAINING_CONFIG = {
+ "learning_rate": 5e-5,
+ "batch_size": 4,
+ "gradient_accumulation_steps": 8,
+ "num_epochs": 50,
+ "warmup_steps": 1000,
+ "scheduler": "cosine_with_restarts",
+ "optimizer": "adamw",
+ "weight_decay": 0.01,
+ "mixed_precision": True,
+ "gradient_clipping": 1.0,
+}
+
+# Data Configuration
+DATA_CONFIG = {
+ "dataset_size": 1000,
+ "train_split": 0.8,
+ "val_split": 0.15,
+ "test_split": 0.05,
+ "sample_rate": 48000,
+ "bit_depth": 24,
+ "min_duration": 30,
+ "max_duration": 180,
+}
+
+# Audio Processing
+AUDIO_CONFIG = {
+ "n_fft": 2048,
+ "hop_length": 512,
+ "n_mels": 128,
+ "fmin": 20,
+ "fmax": 8000,
+ "window": "hann",
+}
+
+# Style Definitions
+STYLE_DEFINITIONS = {
+ "lam_plearn": {
+ "thai_name": "หมอลำเพลิน",
+ "description": "Slow, contemplative style",
+ "tempo_range": [60, 90],
+ "mood": ["sad", "romantic", "contemplative"],
+ },
+ "lam_sing": {
+ "thai_name": "หมอลำซิ่ง",
+ "description": "Fast, energetic style",
+ "tempo_range": [120, 160],
+ "mood": ["fun", "exciting", "energetic"],
+ },
+ "lam_klorn": {
+ "thai_name": "หมอลำกลอน",
+ "description": "Poetic, storytelling style",
+ "tempo_range": [80, 120],
+ "mood": ["romantic", "dramatic", "nostalgic"],
+ },
+ "lam_tad": {
+ "thai_name": "ลำตัด",
+ "description": "Cutting, abrupt style",
+ "tempo_range": [100, 140],
+ "mood": ["dramatic", "intense", "powerful"],
+ },
+ "lam_puen": {
+ "thai_name": "ลำพื้น",
+ "description": "Traditional, grounded style",
+ "tempo_range": [70, 110],
+ "mood": ["traditional", "authentic", "earthy"],
+ },
+}
+
+# Regional Styles
+REGIONAL_STYLES = {
+ "northeast_north": {
+ "thai_name": "อีสานเหนือ",
+ "characteristics": ["melodic", "ornamented", "graceful"],
+ "instruments": ["pin", "khaen", "pong lang"],
+ },
+ "northeast_south": {
+ "thai_name": "อีสานใต้",
+ "characteristics": ["rhythmic", "energetic", "driving"],
+ "instruments": ["pin", "khaen", "drums"],
+ },
+ "northeast_central": {
+ "thai_name": "อีสานกลาง",
+ "characteristics": ["balanced", "versatile", "expressive"],
+ "instruments": ["pin", "khaen", "pong lang", "drums"],
+ },
+}
+
+# Mood Definitions
+MOOD_DEFINITIONS = {
+ "sad": {
+ "thai_name": "เศร้า",
+ "description": "Melancholic, emotional, touching",
+ "musical_features": ["minor_key", "slow_tempo", "descending_melodies"],
+ },
+ "fun": {
+ "thai_name": "สนุกสนาน",
+ "description": "Joyful, entertaining, lively",
+ "musical_features": ["major_key", "fast_tempo", "ascending_melodies"],
+ },
+ "romantic": {
+ "thai_name": "โรแมนติก",
+ "description": "Loving, tender, intimate",
+ "musical_features": ["soft_dynamics", "legato", "expressive_vibrato"],
+ },
+ "exciting": {
+ "thai_name": "ตื่นเต้น",
+ "description": "Thrilling, energetic, intense",
+ "musical_features": ["dynamics_changes", "syncopation", "ornamentation"],
+ },
+ "contemplative": {
+ "thai_name": "ครุ่นคิด",
+ "description": "Thoughtful, meditative, reflective",
+ "musical_features": ["slow_tempo", "sustained_notes", "minimal_ornamentation"],
+ },
+}
+
+# Technical Parameters
+TECHNICAL_CONFIG = {
+ "device": "cuda",
+ "mixed_precision": True,
+ "compile_model": False,
+ "torch_compile_mode": "default",
+ "seed": 42,
+ "deterministic": True,
+}
+
+# Evaluation Metrics
+EVALUATION_CONFIG = {
+ "objective_metrics": ["FAD", "MCD", "SNR", "STFT_loss"],
+ "subjective_metrics": ["authenticity", "musicality", "style_accuracy", "satisfaction"],
+ "human_evaluation": True,
+ "expert_evaluation": True,
+ "listening_tests": True,
+}
+
+# Web Application
+WEB_CONFIG = {
+ "host": "0.0.0.0",
+ "port": 8000,
+ "debug": False,
+ "max_file_size": 50 * 1024 * 1024, # 50MB
+ "allowed_extensions": [".wav", ".mp3", ".flac", ".m4a"],
+ "rate_limit": 10, # requests per minute
+}
+
+# API Configuration
+API_CONFIG = {
+ "version": "v1",
+ "title": "Isan Pin AI API",
+ "description": "API for generating traditional Isan Pin music using AI",
+ "docs_url": "/docs",
+ "redoc_url": "/redoc",
+}
+
+# Storage Configuration
+STORAGE_CONFIG = {
+ "audio_dir": "data/audio",
+ "models_dir": "models",
+ "logs_dir": "logs",
+ "temp_dir": "temp",
+ "cache_dir": "cache",
+}
+
+# Logging Configuration
+LOGGING_CONFIG = {
+ "level": "INFO",
+ "format": "%(asctime)s - %(name)s - %(levelname)s - %(message)s",
+ "file_logging": True,
+ "console_logging": True,
+ "max_file_size": 10 * 1024 * 1024, # 10MB
+ "backup_count": 5,
+}
+
+# Security Configuration
+SECURITY_CONFIG = {
+ "api_key_required": False,
+ "rate_limiting": True,
+ "file_validation": True,
+ "content_filtering": True,
+ "audit_logging": True,
+}
\ No newline at end of file
diff --git a/src/data/collection.py b/src/data/collection.py
new file mode 100644
index 000000000000..e915da703cff
--- /dev/null
+++ b/src/data/collection.py
@@ -0,0 +1,523 @@
+"""
+Data Collection Module for Isan Pin AI
+
+This module handles:
+- Audio file collection and organization
+- Metadata extraction and management
+- Audio format conversion and standardization
+- Quality assessment and filtering
+"""
+
+import os
+import json
+import logging
+import librosa
+import soundfile as sf
+import numpy as np
+import pandas as pd
+from pathlib import Path
+from typing import List, Dict, Optional, Tuple
+import torch
+import torchaudio
+from datetime import datetime
+
+from ..config import DATA_CONFIG, AUDIO_CONFIG, STORAGE_CONFIG, LOGGING_CONFIG
+from ..utils import AudioUtils, TextProcessor
+
+logger = logging.getLogger(__name__)
+
+class DataCollector:
+ """
+ Collects and preprocesses audio data for Isan Pin AI training
+ """
+
+ def __init__(
+ self,
+ input_dir: str,
+ output_dir: str,
+ sample_rate: int = None,
+ classifier: Optional['AudioClassifier'] = None
+ ):
+ """
+ Initialize the data collector
+
+ Args:
+ input_dir: Directory containing raw audio files
+ output_dir: Directory for processed output
+ sample_rate: Target sample rate (uses config default if None)
+ classifier: Optional AudioClassifier for automatic classification
+ """
+ self.input_dir = Path(input_dir)
+ self.output_dir = Path(output_dir)
+ self.sample_rate = sample_rate or DATA_CONFIG["sample_rate"]
+ self.classifier = classifier
+
+ # Create output directories
+ self.audio_output_dir = self.output_dir / "audio"
+ self.metadata_dir = self.output_dir / "metadata"
+ self.temp_dir = self.output_dir / "temp"
+
+ for dir_path in [self.audio_output_dir, self.metadata_dir, self.temp_dir]:
+ dir_path.mkdir(parents=True, exist_ok=True)
+
+ self.audio_utils = AudioUtils()
+ self.text_processor = TextProcessor()
+
+ logger.info(f"Initialized DataCollector: input={input_dir}, output={output_dir}")
+
+ def process_all(self) -> Dict:
+ """
+ Process all audio files in the input directory
+
+ Returns:
+ Dictionary with processing statistics
+ """
+ logger.info("Starting batch processing of audio files")
+
+ # Find all audio files
+ audio_files = self._find_audio_files()
+ logger.info(f"Found {len(audio_files)} audio files")
+
+ if not audio_files:
+ logger.warning("No audio files found in input directory")
+ return {"processed": 0, "errors": 0, "total": 0}
+
+ # Process files
+ results = []
+ errors = []
+
+ for i, audio_file in enumerate(audio_files, 1):
+ logger.info(f"Processing file {i}/{len(audio_files)}: {audio_file.name}")
+
+ try:
+ result = self._process_single_file(audio_file)
+ results.append(result)
+ logger.info(f"Successfully processed: {audio_file.name}")
+ except Exception as e:
+ errors.append({"file": str(audio_file), "error": str(e)})
+ logger.error(f"Error processing {audio_file.name}: {e}")
+
+ # Save processing log
+ self._save_processing_log(results, errors)
+
+ stats = {
+ "processed": len(results),
+ "errors": len(errors),
+ "total": len(audio_files),
+ "success_rate": len(results) / len(audio_files) * 100,
+ }
+
+ logger.info(f"Processing completed: {stats}")
+ return stats
+
+ def _find_audio_files(self) -> List[Path]:
+ """Find all audio files in the input directory"""
+ audio_extensions = [".wav", ".mp3", ".flac", ".m4a", ".aac", ".ogg"]
+ audio_files = []
+
+ for ext in audio_extensions:
+ audio_files.extend(self.input_dir.rglob(f"*{ext}"))
+ audio_files.extend(self.input_dir.rglob(f"*{ext.upper()}"))
+
+ return sorted(list(set(audio_files))) # Remove duplicates
+
+ def _process_single_file(self, audio_file: Path) -> Dict:
+ """
+ Process a single audio file
+
+ Args:
+ audio_file: Path to the audio file
+
+ Returns:
+ Dictionary with processing results
+ """
+ logger.debug(f"Processing file: {audio_file}")
+
+ # Load audio
+ try:
+ audio, sr = librosa.load(audio_file, sr=self.sample_rate, mono=True)
+ except Exception as e:
+ raise Exception(f"Failed to load audio: {e}")
+
+ # Get audio info
+ duration = len(audio) / self.sample_rate
+
+ # Validate audio
+ if duration < DATA_CONFIG["min_duration"]:
+ raise Exception(f"Audio too short: {duration:.2f}s < {DATA_CONFIG['min_duration']}s")
+
+ if duration > DATA_CONFIG["max_duration"]:
+ logger.warning(f"Audio longer than max: {duration:.2f}s")
+ # Optionally trim or split
+ audio = audio[:int(DATA_CONFIG["max_duration"] * self.sample_rate)]
+ duration = DATA_CONFIG["max_duration"]
+
+ # Quality assessment
+ quality_metrics = self._assess_audio_quality(audio)
+
+ # Generate filename
+ output_filename = self._generate_output_filename(audio_file)
+ output_path = self.audio_output_dir / output_filename
+
+ # Save processed audio
+ sf.write(output_path, audio, self.sample_rate)
+
+ # Extract or generate metadata
+ metadata = self._extract_metadata(audio_file, audio, duration, quality_metrics)
+
+ # Classify if classifier is available
+ if self.classifier:
+ classification = self.classifier.classify(audio)
+ metadata.update(classification)
+
+ # Save metadata
+ metadata_path = self.metadata_dir / f"{output_filename.stem}.json"
+ with open(metadata_path, 'w', encoding='utf-8') as f:
+ json.dump(metadata, f, ensure_ascii=False, indent=2)
+
+ result = {
+ "input_file": str(audio_file),
+ "output_file": str(output_path),
+ "metadata_file": str(metadata_path),
+ "duration": duration,
+ "sample_rate": self.sample_rate,
+ "quality_score": quality_metrics.get("overall_score", 0),
+ "processing_time": datetime.now().isoformat(),
+ }
+
+ return result
+
+ def _assess_audio_quality(self, audio: np.ndarray) -> Dict:
+ """
+ Assess the quality of audio
+
+ Args:
+ audio: Audio signal
+
+ Returns:
+ Dictionary with quality metrics
+ """
+ metrics = {}
+
+ try:
+ # Signal-to-noise ratio (simplified)
+ signal_power = np.mean(audio ** 2)
+ if signal_power > 0:
+ # Estimate noise floor (last 10% of signal)
+ noise_samples = int(len(audio) * 0.1)
+ noise_power = np.mean(audio[-noise_samples:] ** 2)
+ snr = 10 * np.log10(signal_power / (noise_power + 1e-10))
+ metrics["snr"] = float(snr)
+ else:
+ metrics["snr"] = -float('inf')
+
+ # Dynamic range
+ dynamic_range = np.max(audio) - np.min(audio)
+ metrics["dynamic_range"] = float(dynamic_range)
+
+ # Zero crossing rate (for detecting silence)
+ zcr = np.mean(np.abs(np.diff(np.signbit(audio))))
+ metrics["zero_crossing_rate"] = float(zcr)
+
+ # Overall quality score (0-10)
+ # This is a simplified version - in practice, you'd want more sophisticated metrics
+ quality_components = []
+
+ # SNR component
+ if metrics["snr"] > 20:
+ quality_components.append(1.0)
+ elif metrics["snr"] > 10:
+ quality_components.append(0.7)
+ else:
+ quality_components.append(0.3)
+
+ # Dynamic range component
+ if dynamic_range > 0.1:
+ quality_components.append(1.0)
+ elif dynamic_range > 0.05:
+ quality_components.append(0.7)
+ else:
+ quality_components.append(0.3)
+
+ # Zero crossing rate component (lower is generally better for music)
+ if zcr < 0.3:
+ quality_components.append(1.0)
+ elif zcr < 0.5:
+ quality_components.append(0.7)
+ else:
+ quality_components.append(0.3)
+
+ metrics["overall_score"] = float(np.mean(quality_components) * 10)
+
+ except Exception as e:
+ logger.error(f"Error assessing audio quality: {e}")
+ metrics["overall_score"] = 0.0
+
+ return metrics
+
+ def _extract_metadata(self, audio_file: Path, audio: np.ndarray, duration: float, quality_metrics: Dict) -> Dict:
+ """
+ Extract metadata from audio file and signal
+
+ Args:
+ audio_file: Original audio file path
+ audio: Audio signal
+ duration: Duration in seconds
+ quality_metrics: Quality assessment results
+
+ Returns:
+ Dictionary with metadata
+ """
+ metadata = {
+ "filename": audio_file.name,
+ "original_path": str(audio_file),
+ "duration": duration,
+ "sample_rate": self.sample_rate,
+ "channels": 1, # Always mono after processing
+ "bit_depth": DATA_CONFIG["bit_depth"],
+ "file_size": audio_file.stat().st_size,
+ "processing_date": datetime.now().isoformat(),
+ "quality_metrics": quality_metrics,
+ }
+
+ # Extract features using librosa
+ try:
+ # Tempo estimation
+ tempo, beats = librosa.beat.beat_track(y=audio, sr=self.sample_rate)
+ metadata["tempo_bpm"] = float(tempo)
+
+ # Key estimation (simplified)
+ chroma = librosa.feature.chroma_stft(y=audio, sr=self.sample_rate)
+ key_idx = np.argmax(np.mean(chroma, axis=1))
+ keys = ['C', 'C#', 'D', 'D#', 'E', 'F', 'F#', 'G', 'G#', 'A', 'A#', 'B']
+ metadata["estimated_key"] = keys[key_idx]
+
+ # Spectral features
+ spectral_centroid = librosa.feature.spectral_centroid(y=audio, sr=self.sample_rate)
+ metadata["spectral_centroid"] = float(np.mean(spectral_centroid))
+
+ spectral_rolloff = librosa.feature.spectral_rolloff(y=audio, sr=self.sample_rate)
+ metadata["spectral_rolloff"] = float(np.mean(spectral_rolloff))
+
+ zero_crossing_rate = librosa.feature.zero_crossing_rate(audio)
+ metadata["zero_crossing_rate"] = float(np.mean(zero_crossing_rate))
+
+ # MFCC features (first 13 coefficients)
+ mfcc = librosa.feature.mfcc(y=audio, sr=self.sample_rate, n_mfcc=13)
+ metadata["mfcc_mean"] = np.mean(mfcc, axis=1).tolist()
+ metadata["mfcc_std"] = np.std(mfcc, axis=1).tolist()
+
+ except Exception as e:
+ logger.warning(f"Could not extract audio features: {e}")
+
+ # Add file metadata
+ stat = audio_file.stat()
+ metadata["file_metadata"] = {
+ "created": datetime.fromtimestamp(stat.st_ctime).isoformat(),
+ "modified": datetime.fromtimestamp(stat.st_mtime).isoformat(),
+ "size_bytes": stat.st_size,
+ "size_mb": round(stat.st_size / (1024 * 1024), 2),
+ }
+
+ return metadata
+
+ def _generate_output_filename(self, input_file: Path) -> str:
+ """Generate output filename based on input file"""
+ timestamp = datetime.now().strftime("%Y%m%d_%H%M%S")
+ base_name = input_file.stem.replace(" ", "_").replace("-", "_")
+ return f"{base_name}_{timestamp}.wav"
+
+ def _save_processing_log(self, results: List[Dict], errors: List[Dict]):
+ """Save processing log to file"""
+ log_data = {
+ "processing_date": datetime.now().isoformat(),
+ "input_directory": str(self.input_dir),
+ "output_directory": str(self.output_dir),
+ "sample_rate": self.sample_rate,
+ "total_files": len(results) + len(errors),
+ "successful": len(results),
+ "errors": len(errors),
+ "results": results,
+ "errors": errors,
+ }
+
+ log_path = self.output_dir / "processing_log.json"
+ with open(log_path, 'w', encoding='utf-8') as f:
+ json.dump(log_data, f, ensure_ascii=False, indent=2)
+
+ logger.info(f"Processing log saved to: {log_path}")
+
+
+class AudioClassifier:
+ """
+ Classifies Isan Pin music based on audio features
+ """
+
+ def __init__(self):
+ """Initialize the audio classifier"""
+ self.audio_utils = AudioUtils()
+ self.text_processor = TextProcessor()
+ logger.info("Initialized AudioClassifier")
+
+ def classify(self, audio: np.ndarray, sample_rate: int = None) -> Dict:
+ """
+ Classify audio into Isan Pin music categories
+
+ Args:
+ audio: Audio signal
+ sample_rate: Sample rate of the audio
+
+ Returns:
+ Dictionary with classification results
+ """
+ if sample_rate is None:
+ sample_rate = DATA_CONFIG["sample_rate"]
+
+ try:
+ # Extract audio features
+ features = self._extract_classification_features(audio, sample_rate)
+
+ # Classify based on features
+ classification = self._classify_features(features)
+
+ return classification
+
+ except Exception as e:
+ logger.error(f"Error classifying audio: {e}")
+ return {
+ "style": "unknown",
+ "confidence": 0.0,
+ "reasoning": f"Classification failed: {e}",
+ }
+
+ def _extract_classification_features(self, audio: np.ndarray, sample_rate: int) -> Dict:
+ """Extract features for classification"""
+ features = {}
+
+ # Tempo estimation
+ tempo, _ = librosa.beat.beat_track(y=audio, sr=sample_rate)
+ features["tempo"] = float(tempo)
+
+ # Spectral features
+ spectral_centroid = librosa.feature.spectral_centroid(y=audio, sr=sample_rate)
+ features["spectral_centroid_mean"] = float(np.mean(spectral_centroid))
+ features["spectral_centroid_std"] = float(np.std(spectral_centroid))
+
+ spectral_bandwidth = librosa.feature.spectral_bandwidth(y=audio, sr=sample_rate)
+ features["spectral_bandwidth_mean"] = float(np.mean(spectral_bandwidth))
+
+ spectral_rolloff = librosa.feature.spectral_rolloff(y=audio, sr=sample_rate)
+ features["spectral_rolloff_mean"] = float(np.mean(spectral_rolloff))
+
+ # Zero crossing rate
+ zcr = librosa.feature.zero_crossing_rate(audio)
+ features["zcr_mean"] = float(np.mean(zcr))
+ features["zcr_std"] = float(np.std(zcr))
+
+ # MFCC features
+ mfcc = librosa.feature.mfcc(y=audio, sr=sample_rate, n_mfcc=13)
+ features["mfcc_means"] = np.mean(mfcc, axis=1).tolist()
+ features["mfcc_stds"] = np.std(mfcc, axis=1).tolist()
+
+ # Chroma features
+ chroma = librosa.feature.chroma_stft(y=audio, sr=sample_rate)
+ features["chroma_means"] = np.mean(chroma, axis=1).tolist()
+
+ # Spectral contrast
+ contrast = librosa.feature.spectral_contrast(y=audio, sr=sample_rate)
+ features["contrast_means"] = np.mean(contrast, axis=1).tolist()
+
+ # Tonnetz
+ tonnetz = librosa.feature.tonnetz(y=audio, sr=sample_rate)
+ features["tonnetz_means"] = np.mean(tonnetz, axis=1).tolist()
+
+ # Rhythm features
+ onset_env = librosa.onset.onset_strength(y=audio, sr=sample_rate)
+ tempo = librosa.beat.tempo(onset_envelope=onset_env, sr=sample_rate)[0]
+ features["tempo_estimated"] = float(tempo)
+
+ # Dynamic features
+ rms = librosa.feature.rms(y=audio)
+ features["rms_mean"] = float(np.mean(rms))
+ features["rms_std"] = float(np.std(rms))
+
+ return features
+
+ def _classify_features(self, features: Dict) -> Dict:
+ """Classify audio based on extracted features"""
+ # This is a simplified rule-based classifier
+ # In a real implementation, you would use a trained ML model
+
+ tempo = features.get("tempo", 0)
+ spectral_centroid = features.get("spectral_centroid_mean", 0)
+ zcr = features.get("zcr_mean", 0)
+ rms_mean = features.get("rms_mean", 0)
+
+ # Rule-based classification
+ classification = {
+ "style": "unknown",
+ "confidence": 0.5,
+ "reasoning": "Rule-based classification",
+ }
+
+ # Style classification based on tempo and spectral features
+ if tempo < 80:
+ classification["style"] = "lam_plearn"
+ classification["confidence"] = 0.7
+ classification["reasoning"] = f"Slow tempo ({tempo:.1f} BPM) suggests Lam Plearn style"
+ elif tempo > 120:
+ classification["style"] = "lam_sing"
+ classification["confidence"] = 0.7
+ classification["reasoning"] = f"Fast tempo ({tempo:.1f} BPM) suggests Lam Sing style"
+ elif 80 <= tempo <= 120:
+ if spectral_centroid > 2000:
+ classification["style"] = "lam_klorn"
+ classification["confidence"] = 0.6
+ classification["reasoning"] = f"Medium tempo with high spectral centroid suggests Lam Klorn"
+ else:
+ classification["style"] = "lam_tad"
+ classification["confidence"] = 0.6
+ classification["reasoning"] = f"Medium tempo with lower spectral centroid suggests Lam Tad"
+
+ # Add mood estimation
+ if rms_mean < 0.05:
+ classification["mood"] = "sad"
+ classification["mood_confidence"] = 0.6
+ elif rms_mean > 0.15:
+ classification["mood"] = "fun"
+ classification["mood_confidence"] = 0.6
+ else:
+ classification["mood"] = "contemplative"
+ classification["mood_confidence"] = 0.5
+
+ # Add tempo category
+ if tempo < 80:
+ classification["tempo_category"] = "slow"
+ elif tempo > 120:
+ classification["tempo_category"] = "fast"
+ else:
+ classification["tempo_category"] = "medium"
+
+ # Add key estimation (simplified)
+ # This would be more sophisticated in a real implementation
+ classification["estimated_key"] = "D" # Common key for Isan music
+ classification["key_confidence"] = 0.4
+
+ return classification
+
+
+if __name__ == "__main__":
+ # Example usage
+ collector = DataCollector(
+ input_dir="./raw_audio",
+ output_dir="./processed_data",
+ sample_rate=48000,
+ )
+
+ # Add classifier
+ classifier = AudioClassifier()
+ collector.classifier = classifier
+
+ # Process all files
+ stats = collector.process_all()
+ print(f"Processing completed: {stats}")
\ No newline at end of file
diff --git a/src/data/preprocessing.py b/src/data/preprocessing.py
new file mode 100644
index 000000000000..0f0940f13dd1
--- /dev/null
+++ b/src/data/preprocessing.py
@@ -0,0 +1,590 @@
+"""
+Audio Preprocessing Module for Isan Pin AI
+
+This module handles:
+- Audio format standardization
+- Noise reduction and filtering
+- Data augmentation
+- Feature extraction
+- Dataset creation for training
+"""
+
+import os
+import json
+import random
+import logging
+import numpy as np
+import librosa
+import soundfile as sf
+import torch
+import torchaudio
+from pathlib import Path
+from typing import List, Dict, Tuple, Optional, Callable
+from sklearn.model_selection import train_test_split
+from sklearn.preprocessing import LabelEncoder
+import pandas as pd
+
+from ..config import DATA_CONFIG, AUDIO_CONFIG, STORAGE_CONFIG
+from ..utils import AudioUtils
+
+logger = logging.getLogger(__name__)
+
+class AudioPreprocessor:
+ """
+ Preprocesses audio data for machine learning training
+ """
+
+ def __init__(
+ self,
+ sample_rate: int = None,
+ n_fft: int = None,
+ hop_length: int = None,
+ n_mels: int = None,
+ ):
+ """
+ Initialize the audio preprocessor
+
+ Args:
+ sample_rate: Target sample rate
+ n_fft: FFT window size
+ hop_length: Hop length for STFT
+ n_mels: Number of mel bands
+ """
+ self.sample_rate = sample_rate or DATA_CONFIG["sample_rate"]
+ self.n_fft = n_fft or AUDIO_CONFIG["n_fft"]
+ self.hop_length = hop_length or AUDIO_CONFIG["hop_length"]
+ self.n_mels = n_mels or AUDIO_CONFIG["n_mels"]
+
+ self.audio_utils = AudioUtils()
+
+ logger.info(f"Initialized AudioPreprocessor: sr={self.sample_rate}, n_fft={self.n_fft}")
+
+ def standardize_audio(self, audio: np.ndarray, target_sr: int = None) -> np.ndarray:
+ """
+ Standardize audio format
+
+ Args:
+ audio: Input audio signal
+ target_sr: Target sample rate (uses default if None)
+
+ Returns:
+ Standardized audio signal
+ """
+ target_sr = target_sr or self.sample_rate
+
+ # Convert to mono if stereo
+ if len(audio.shape) > 1:
+ audio = librosa.to_mono(audio)
+
+ # Resample if necessary
+ if target_sr != self.sample_rate:
+ audio = librosa.resample(audio, orig_sr=self.sample_rate, target_sr=target_sr)
+
+ # Normalize
+ audio = librosa.util.normalize(audio)
+
+ return audio
+
+ def extract_features(self, audio: np.ndarray, feature_type: str = "all") -> Dict:
+ """
+ Extract audio features for analysis and classification
+
+ Args:
+ audio: Audio signal
+ feature_type: Type of features to extract ('all', 'basic', 'spectral', 'temporal')
+
+ Returns:
+ Dictionary with extracted features
+ """
+ features = {}
+
+ if feature_type in ["all", "basic"]:
+ # Basic features
+ features["duration"] = len(audio) / self.sample_rate
+ features["mean"] = float(np.mean(audio))
+ features["std"] = float(np.std(audio))
+ features["rms"] = float(np.sqrt(np.mean(audio ** 2)))
+ features["zero_crossing_rate"] = float(np.mean(librosa.feature.zero_crossing_rate(audio)))
+
+ if feature_type in ["all", "spectral"]:
+ # Spectral features
+ spectral_centroid = librosa.feature.spectral_centroid(y=audio, sr=self.sample_rate)
+ features["spectral_centroid_mean"] = float(np.mean(spectral_centroid))
+ features["spectral_centroid_std"] = float(np.std(spectral_centroid))
+
+ spectral_bandwidth = librosa.feature.spectral_bandwidth(y=audio, sr=self.sample_rate)
+ features["spectral_bandwidth_mean"] = float(np.mean(spectral_bandwidth))
+
+ spectral_rolloff = librosa.feature.spectral_rolloff(y=audio, sr=self.sample_rate)
+ features["spectral_rolloff_mean"] = float(np.mean(spectral_rolloff))
+
+ spectral_contrast = librosa.feature.spectral_contrast(y=audio, sr=self.sample_rate)
+ features["spectral_contrast_mean"] = np.mean(spectral_contrast, axis=1).tolist()
+
+ # MFCC features
+ mfcc = librosa.feature.mfcc(y=audio, sr=self.sample_rate, n_mfcc=13)
+ features["mfcc_means"] = np.mean(mfcc, axis=1).tolist()
+ features["mfcc_stds"] = np.std(mfcc, axis=1).tolist()
+
+ # Chroma features
+ chroma = librosa.feature.chroma_stft(y=audio, sr=self.sample_rate)
+ features["chroma_means"] = np.mean(chroma, axis=1).tolist()
+
+ if feature_type in ["all", "temporal"]:
+ # Temporal features
+ tempo, beats = librosa.beat.beat_track(y=audio, sr=self.sample_rate)
+ features["tempo"] = float(tempo)
+ features["beats_per_second"] = len(beats) / features["duration"] if features.get("duration") else 0
+
+ # Onset detection
+ onset_env = librosa.onset.onset_strength(y=audio, sr=self.sample_rate)
+ features["onset_strength_mean"] = float(np.mean(onset_env))
+
+ # RMS energy
+ rms = librosa.feature.rms(y=audio)
+ features["rms_mean"] = float(np.mean(rms))
+ features["rms_std"] = float(np.std(rms))
+
+ return features
+
+ def create_spectrogram(self, audio: np.ndarray, save_path: str = None) -> np.ndarray:
+ """
+ Create mel-spectrogram from audio
+
+ Args:
+ audio: Audio signal
+ save_path: Optional path to save the spectrogram image
+
+ Returns:
+ Mel-spectrogram as numpy array
+ """
+ # Compute mel-spectrogram
+ mel_spec = librosa.feature.melspectrogram(
+ y=audio,
+ sr=self.sample_rate,
+ n_fft=self.n_fft,
+ hop_length=self.hop_length,
+ n_mels=self.n_mels,
+ )
+
+ # Convert to log scale
+ mel_spec_db = librosa.power_to_db(mel_spec, ref=np.max)
+
+ # Save as image if requested
+ if save_path:
+ import matplotlib.pyplot as plt
+ plt.figure(figsize=(12, 4))
+ librosa.display.specshow(mel_spec_db, sr=self.sample_rate, hop_length=self.hop_length, x_axis='time', y_axis='mel')
+ plt.colorbar(format='%+2.0f dB')
+ plt.title('Mel Spectrogram')
+ plt.tight_layout()
+ plt.savefig(save_path)
+ plt.close()
+
+ return mel_spec_db
+
+ def apply_noise_reduction(self, audio: np.ndarray, noise_reduction_factor: float = 0.5) -> np.ndarray:
+ """
+ Apply simple noise reduction
+
+ Args:
+ audio: Input audio
+ noise_reduction_factor: Strength of noise reduction (0.0-1.0)
+
+ Returns:
+ Noise-reduced audio
+ """
+ # Simple spectral subtraction noise reduction
+ # This is a basic implementation - for production, use more sophisticated methods
+
+ try:
+ # Compute STFT
+ stft = librosa.stft(audio, n_fft=self.n_fft, hop_length=self.hop_length)
+ magnitude = np.abs(stft)
+ phase = np.angle(stft)
+
+ # Estimate noise profile from first few frames
+ noise_profile = np.mean(magnitude[:, :5], axis=1, keepdims=True)
+
+ # Apply spectral subtraction
+ magnitude_denoised = magnitude - noise_reduction_factor * noise_profile
+ magnitude_denoised = np.maximum(magnitude_denoised, 0)
+
+ # Reconstruct signal
+ stft_denoised = magnitude_denoised * np.exp(1j * phase)
+ audio_denoised = librosa.istft(stft_denoised, hop_length=self.hop_length)
+
+ return audio_denoised
+
+ except Exception as e:
+ logger.warning(f"Noise reduction failed: {e}. Returning original audio.")
+ return audio
+
+ def apply_filtering(self, audio: np.ndarray, filter_type: str = "highpass", cutoff_freq: float = 80.0) -> np.ndarray:
+ """
+ Apply audio filtering
+
+ Args:
+ audio: Input audio
+ filter_type: Type of filter ('highpass', 'lowpass', 'bandpass')
+ cutoff_freq: Cutoff frequency in Hz
+
+ Returns:
+ Filtered audio
+ """
+ try:
+ from scipy import signal
+
+ # Design filter
+ nyquist = self.sample_rate / 2
+
+ if filter_type == "highpass":
+ sos = signal.butter(4, cutoff_freq / nyquist, btype='high', output='sos')
+ elif filter_type == "lowpass":
+ sos = signal.butter(4, cutoff_freq / nyquist, btype='low', output='sos')
+ elif filter_type == "bandpass":
+ low_cutoff = cutoff_freq[0] / nyquist
+ high_cutoff = cutoff_freq[1] / nyquist
+ sos = signal.butter(4, [low_cutoff, high_cutoff], btype='band', output='sos')
+ else:
+ raise ValueError(f"Unknown filter type: {filter_type}")
+
+ # Apply filter
+ filtered_audio = signal.sosfilt(sos, audio)
+
+ return filtered_audio
+
+ except Exception as e:
+ logger.warning(f"Filtering failed: {e}. Returning original audio.")
+ return audio
+
+ def augment_audio(self, audio: np.ndarray, augmentation_type: str = "pitch_shift", **kwargs) -> np.ndarray:
+ """
+ Apply data augmentation to audio
+
+ Args:
+ audio: Input audio
+ augmentation_type: Type of augmentation
+ **kwargs: Additional parameters for augmentation
+
+ Returns:
+ Augmented audio
+ """
+ try:
+ if augmentation_type == "pitch_shift":
+ n_steps = kwargs.get("n_steps", random.uniform(-2, 2))
+ return librosa.effects.pitch_shift(audio, sr=self.sample_rate, n_steps=n_steps)
+
+ elif augmentation_type == "time_stretch":
+ rate = kwargs.get("rate", random.uniform(0.9, 1.1))
+ return librosa.effects.time_stretch(audio, rate=rate)
+
+ elif augmentation_type == "add_noise":
+ noise_factor = kwargs.get("noise_factor", random.uniform(0.01, 0.05))
+ noise = np.random.normal(0, noise_factor, len(audio))
+ return audio + noise
+
+ elif augmentation_type == "volume_change":
+ gain = kwargs.get("gain", random.uniform(0.7, 1.3))
+ return audio * gain
+
+ elif augmentation_type == "reverb":
+ # Simple reverb effect
+ delay_samples = int(kwargs.get("delay_ms", 100) * self.sample_rate / 1000)
+ decay = kwargs.get("decay", 0.5)
+
+ reverb = np.zeros_like(audio)
+ if len(audio) > delay_samples:
+ reverb[delay_samples:] = audio[:-delay_samples] * decay
+
+ return audio + reverb
+
+ else:
+ raise ValueError(f"Unknown augmentation type: {augmentation_type}")
+
+ except Exception as e:
+ logger.warning(f"Augmentation failed: {e}. Returning original audio.")
+ return audio
+
+
+class DatasetCreator:
+ """
+ Creates training datasets for Isan Pin AI
+ """
+
+ def __init__(
+ self,
+ output_dir: str,
+ preprocessor: Optional[AudioPreprocessor] = None,
+ ):
+ """
+ Initialize dataset creator
+
+ Args:
+ output_dir: Output directory for dataset
+ preprocessor: Audio preprocessor instance
+ """
+ self.output_dir = Path(output_dir)
+ self.preprocessor = preprocessor or AudioPreprocessor()
+
+ # Create output directories
+ self.train_dir = self.output_dir / "train"
+ self.val_dir = self.output_dir / "validation"
+ self.test_dir = self.output_dir / "test"
+
+ for dir_path in [self.train_dir, self.val_dir, self.test_dir]:
+ dir_path.mkdir(parents=True, exist_ok=True)
+
+ logger.info(f"Initialized DatasetCreator: output={output_dir}")
+
+ def create_dataset(
+ self,
+ audio_files: List[Path],
+ labels: Optional[List[str]] = None,
+ test_size: float = 0.15,
+ val_size: float = 0.05,
+ random_state: int = 42,
+ augment_train: bool = True,
+ ) -> Dict:
+ """
+ Create training dataset from audio files
+
+ Args:
+ audio_files: List of audio file paths
+ labels: Optional labels for supervised learning
+ test_size: Proportion of data for testing
+ val_size: Proportion of data for validation
+ random_state: Random seed for reproducibility
+ augment_train: Whether to augment training data
+
+ Returns:
+ Dictionary with dataset information
+ """
+ logger.info(f"Creating dataset from {len(audio_files)} audio files")
+
+ # Split data
+ if labels is not None:
+ # Stratified split if labels are provided
+ X_train, X_temp, y_train, y_temp = train_test_split(
+ audio_files, labels, test_size=(test_size + val_size),
+ random_state=random_state, stratify=labels
+ )
+ X_val, X_test, y_val, y_test = train_test_split(
+ X_temp, y_temp, test_size=test_size/(test_size + val_size),
+ random_state=random_state, stratify=y_temp
+ )
+ else:
+ # Random split
+ X_train, X_temp = train_test_split(
+ audio_files, test_size=(test_size + val_size),
+ random_state=random_state
+ )
+ X_val, X_test = train_test_split(
+ X_temp, test_size=test_size/(test_size + val_size),
+ random_state=random_state
+ )
+ y_train = y_val = y_test = None
+
+ logger.info(f"Split: train={len(X_train)}, val={len(X_val)}, test={len(X_test)}")
+
+ # Process and save each split
+ splits = [
+ ("train", X_train, y_train, self.train_dir, augment_train),
+ ("validation", X_val, y_val, self.val_dir, False),
+ ("test", X_test, y_test, self.test_dir, False),
+ ]
+
+ dataset_info = {}
+
+ for split_name, files, labels, output_dir, augment in splits:
+ logger.info(f"Processing {split_name} split...")
+
+ split_info = self._process_split(
+ files, labels, output_dir, augment, split_name
+ )
+
+ dataset_info[split_name] = split_info
+
+ # Save dataset metadata
+ metadata = {
+ "created_date": pd.Timestamp.now().isoformat(),
+ "total_files": len(audio_files),
+ "train_size": len(X_train),
+ "val_size": len(X_val),
+ "test_size": len(X_test),
+ "test_percentage": test_size * 100,
+ "val_percentage": val_size * 100,
+ "augment_train": augment_train,
+ "sample_rate": self.preprocessor.sample_rate,
+ "splits": dataset_info,
+ }
+
+ metadata_path = self.output_dir / "dataset_metadata.json"
+ with open(metadata_path, 'w', encoding='utf-8') as f:
+ json.dump(metadata, f, ensure_ascii=False, indent=2)
+
+ logger.info(f"Dataset created successfully: {self.output_dir}")
+ return metadata
+
+ def _process_split(
+ self,
+ files: List[Path],
+ labels: Optional[List[str]],
+ output_dir: Path,
+ augment: bool,
+ split_name: str,
+ ) -> Dict:
+ """Process a single data split"""
+
+ processed_files = []
+ augment_files = []
+
+ for i, file_path in enumerate(files):
+ try:
+ # Load and preprocess audio
+ audio, sr = librosa.load(file_path, sr=self.preprocessor.sample_rate, mono=True)
+ audio = self.preprocessor.standardize_audio(audio)
+
+ # Save original
+ output_filename = f"{split_name}_{i:06d}.wav"
+ output_path = output_dir / output_filename
+ sf.write(output_path, audio, self.preprocessor.sample_rate)
+
+ file_info = {
+ "filename": output_filename,
+ "original_path": str(file_path),
+ "duration": len(audio) / self.preprocessor.sample_rate,
+ "sample_rate": self.preprocessor.sample_rate,
+ }
+
+ if labels is not None:
+ file_info["label"] = labels[i]
+
+ processed_files.append(file_info)
+
+ # Apply augmentation if enabled
+ if augment:
+ augment_files.extend(
+ self._augment_file(audio, output_dir, split_name, i, file_info)
+ )
+
+ except Exception as e:
+ logger.error(f"Error processing {file_path}: {e}")
+
+ split_info = {
+ "files": processed_files,
+ "augmented_files": augment_files,
+ "total_original": len(processed_files),
+ "total_augmented": len(augment_files),
+ "total_files": len(processed_files) + len(augment_files),
+ }
+
+ return split_info
+
+ def _augment_file(
+ self,
+ audio: np.ndarray,
+ output_dir: Path,
+ split_name: str,
+ index: int,
+ original_info: Dict,
+ ) -> List[Dict]:
+ """Apply data augmentation to a file"""
+
+ augmented_files = []
+ augmentation_types = ["pitch_shift", "time_stretch", "add_noise"]
+
+ for aug_type in augmentation_types:
+ try:
+ # Apply augmentation
+ augmented_audio = self.preprocessor.augment_audio(audio, aug_type)
+
+ # Save augmented version
+ aug_filename = f"{split_name}_{index:06d}_{aug_type}.wav"
+ aug_path = output_dir / aug_filename
+ sf.write(aug_path, augmented_audio, self.preprocessor.sample_rate)
+
+ aug_info = {
+ "filename": aug_filename,
+ "original_index": index,
+ "augmentation": aug_type,
+ "original_file": original_info["filename"],
+ "duration": len(augmented_audio) / self.preprocessor.sample_rate,
+ }
+
+ if "label" in original_info:
+ aug_info["label"] = original_info["label"]
+
+ augmented_files.append(aug_info)
+
+ except Exception as e:
+ logger.warning(f"Augmentation {aug_type} failed for index {index}: {e}")
+
+ return augmented_files
+
+ def create_metadata_csv(self, dataset_dir: str = None) -> str:
+ """
+ Create CSV metadata file for the dataset
+
+ Args:
+ dataset_dir: Dataset directory (uses self.output_dir if None)
+
+ Returns:
+ Path to the created CSV file
+ """
+ dataset_dir = dataset_dir or self.output_dir
+ dataset_path = Path(dataset_dir)
+
+ all_metadata = []
+
+ # Process each split
+ for split_name in ["train", "validation", "test"]:
+ split_dir = dataset_path / split_name
+ metadata_file = dataset_path / f"{split_name}_metadata.json"
+
+ if metadata_file.exists():
+ with open(metadata_file, 'r', encoding='utf-8') as f:
+ split_data = json.load(f)
+
+ # Add split information to each file
+ for file_info in split_data["files"]:
+ file_info["split"] = split_name
+ file_info["augmented"] = False
+ all_metadata.append(file_info)
+
+ # Add augmented files if they exist
+ if "augmented_files" in split_data:
+ for aug_info in split_data["augmented_files"]:
+ aug_info["split"] = split_name
+ aug_info["augmented"] = True
+ all_metadata.append(aug_info)
+
+ # Create DataFrame
+ df = pd.DataFrame(all_metadata)
+
+ # Save to CSV
+ csv_path = dataset_path / "dataset_metadata.csv"
+ df.to_csv(csv_path, index=False, encoding='utf-8')
+
+ logger.info(f"Metadata CSV created: {csv_path}")
+ return str(csv_path)
+
+
+if __name__ == "__main__":
+ # Example usage
+ preprocessor = AudioPreprocessor()
+
+ # Test audio preprocessing
+ test_audio = np.random.randn(48000) # 1 second of noise
+
+ # Extract features
+ features = preprocessor.extract_features(test_audio)
+ print("Extracted features:", list(features.keys()))
+
+ # Create spectrogram
+ spec = preprocessor.create_spectrogram(test_audio)
+ print("Spectrogram shape:", spec.shape)
+
+ # Test augmentation
+ augmented = preprocessor.augment_audio(test_audio, "pitch_shift", n_steps=2)
+ print("Augmentation successful")
\ No newline at end of file
diff --git a/src/models/__init__.py b/src/models/__init__.py
new file mode 100644
index 000000000000..d8cfe8af6431
--- /dev/null
+++ b/src/models/__init__.py
@@ -0,0 +1 @@
+# Models package
\ No newline at end of file
diff --git a/src/utils/__init__.py b/src/utils/__init__.py
new file mode 100644
index 000000000000..573888ebc8f0
--- /dev/null
+++ b/src/utils/__init__.py
@@ -0,0 +1,407 @@
+"""
+Utility functions for Isan Pin AI
+
+This module provides:
+- Audio processing utilities
+- Text processing utilities
+- Helper functions for the main modules
+"""
+
+import os
+import logging
+import numpy as np
+import librosa
+import soundfile as sf
+from pathlib import Path
+from typing import Union, List, Tuple, Optional, Dict
+
+logger = logging.getLogger(__name__)
+
+
+class AudioUtils:
+ """Utility class for audio processing"""
+
+ def __init__(self):
+ pass
+
+ def load_audio(self, file_path: Union[str, Path], sample_rate: int = None, mono: bool = True) -> Tuple[np.ndarray, int]:
+ """
+ Load audio file with error handling
+
+ Args:
+ file_path: Path to audio file
+ sample_rate: Target sample rate (None for original)
+ mono: Whether to convert to mono
+
+ Returns:
+ Tuple of (audio_signal, sample_rate)
+ """
+ try:
+ audio, sr = librosa.load(file_path, sr=sample_rate, mono=mono)
+ return audio, sr
+ except Exception as e:
+ logger.error(f"Failed to load audio {file_path}: {e}")
+ raise
+
+ def save_audio(self, audio: np.ndarray, file_path: Union[str, Path], sample_rate: int, format: str = "wav"):
+ """
+ Save audio file
+
+ Args:
+ audio: Audio signal
+ file_path: Output file path
+ sample_rate: Sample rate
+ format: Audio format
+ """
+ try:
+ Path(file_path).parent.mkdir(parents=True, exist_ok=True)
+ sf.write(file_path, audio, sample_rate, format=format)
+ logger.info(f"Saved audio: {file_path}")
+ except Exception as e:
+ logger.error(f"Failed to save audio {file_path}: {e}")
+ raise
+
+ def get_audio_info(self, file_path: Union[str, Path]) -> Dict:
+ """
+ Get audio file information
+
+ Args:
+ file_path: Path to audio file
+
+ Returns:
+ Dictionary with audio information
+ """
+ try:
+ info = sf.info(file_path)
+ return {
+ "duration": info.duration,
+ "sample_rate": info.samplerate,
+ "channels": info.channels,
+ "format": info.format,
+ "subtype": info.subtype,
+ "frames": info.frames,
+ }
+ except Exception as e:
+ logger.error(f"Failed to get audio info for {file_path}: {e}")
+ return {}
+
+ def normalize_audio(self, audio: np.ndarray, target_rms: float = 0.1) -> np.ndarray:
+ """
+ Normalize audio to target RMS level
+
+ Args:
+ audio: Input audio
+ target_rms: Target RMS level
+
+ Returns:
+ Normalized audio
+ """
+ current_rms = np.sqrt(np.mean(audio ** 2))
+ if current_rms > 0:
+ scaling_factor = target_rms / current_rms
+ return audio * scaling_factor
+ return audio
+
+ def trim_silence(self, audio: np.ndarray, sample_rate: int, threshold: float = 0.01, frame_length: int = 2048) -> np.ndarray:
+ """
+ Trim silence from beginning and end
+
+ Args:
+ audio: Input audio
+ sample_rate: Sample rate
+ threshold: Silence threshold
+ frame_length: Frame length for analysis
+
+ Returns:
+ Trimmed audio
+ """
+ try:
+ # Find non-silent regions
+ intervals = librosa.effects.split(audio, top_db=20, frame_length=frame_length, hop_length=frame_length//4)
+
+ if len(intervals) > 0:
+ start, end = intervals[0][0], intervals[-1][1]
+ return audio[start:end]
+ else:
+ return audio
+
+ except Exception as e:
+ logger.warning(f"Silence trimming failed: {e}")
+ return audio
+
+ def apply_fade(self, audio: np.ndarray, sample_rate: int, fade_in: float = 0.0, fade_out: float = 0.0) -> np.ndarray:
+ """
+ Apply fade in/out to audio
+
+ Args:
+ audio: Input audio
+ sample_rate: Sample rate
+ fade_in: Fade in duration in seconds
+ fade_out: Fade out duration in seconds
+
+ Returns:
+ Audio with fade applied
+ """
+ try:
+ audio_faded = audio.copy()
+
+ # Apply fade in
+ if fade_in > 0:
+ fade_in_samples = int(fade_in * sample_rate)
+ fade_in_samples = min(fade_in_samples, len(audio_faded))
+ fade_in_curve = np.linspace(0, 1, fade_in_samples)
+ audio_faded[:fade_in_samples] *= fade_in_curve
+
+ # Apply fade out
+ if fade_out > 0:
+ fade_out_samples = int(fade_out * sample_rate)
+ fade_out_samples = min(fade_out_samples, len(audio_faded))
+ fade_out_curve = np.linspace(1, 0, fade_out_samples)
+ audio_faded[-fade_out_samples:] *= fade_out_curve
+
+ return audio_faded
+
+ except Exception as e:
+ logger.warning(f"Fade application failed: {e}")
+ return audio
+
+ def validate_audio(self, audio: np.ndarray, sample_rate: int) -> bool:
+ """
+ Validate audio signal
+
+ Args:
+ audio: Audio signal
+ sample_rate: Sample rate
+
+ Returns:
+ True if valid, False otherwise
+ """
+ try:
+ # Check for NaN or infinite values
+ if np.any(np.isnan(audio)) or np.any(np.isinf(audio)):
+ logger.error("Audio contains NaN or infinite values")
+ return False
+
+ # Check for reasonable amplitude range
+ max_amplitude = np.max(np.abs(audio))
+ if max_amplitude > 10.0:
+ logger.error(f"Audio amplitude too high: {max_amplitude}")
+ return False
+
+ # Check minimum length
+ min_length = sample_rate // 10 # 0.1 seconds minimum
+ if len(audio) < min_length:
+ logger.error(f"Audio too short: {len(audio)} samples")
+ return False
+
+ return True
+
+ except Exception as e:
+ logger.error(f"Audio validation failed: {e}")
+ return False
+
+
+class TextProcessor:
+ """Utility class for text processing"""
+
+ def __init__(self):
+ pass
+
+ def clean_text(self, text: str) -> str:
+ """
+ Clean text input
+
+ Args:
+ text: Input text
+
+ Returns:
+ Cleaned text
+ """
+ if not text:
+ return ""
+
+ # Remove extra whitespace
+ text = " ".join(text.split())
+
+ # Remove special characters (basic cleaning)
+ import re
+ text = re.sub(r'[^\w\sก-๙]', '', text)
+
+ return text.strip()
+
+ def normalize_description(self, description: str, max_length: int = 512) -> str:
+ """
+ Normalize description for model input
+
+ Args:
+ description: Input description
+ max_length: Maximum length
+
+ Returns:
+ Normalized description
+ """
+ description = self.clean_text(description)
+
+ # Truncate if too long
+ words = description.split()
+ if len(words) > max_length // 4: # Rough estimate
+ description = " ".join(words[:max_length // 4])
+
+ return description
+
+ def detect_language(self, text: str) -> str:
+ """
+ Detect language of text (basic detection)
+
+ Args:
+ text: Input text
+
+ Returns:
+ Language code ('th' or 'en')
+ """
+ if not text:
+ return 'en'
+
+ # Simple heuristic: check for Thai characters
+ import re
+ thai_pattern = re.compile(r'[ก-๙]')
+
+ if thai_pattern.search(text):
+ return 'th'
+ else:
+ return 'en'
+
+ def translate_keywords(self, text: str, target_lang: str = 'en') -> str:
+ """
+ Translate key music terms (simplified implementation)
+
+ Args:
+ text: Input text
+ target_lang: Target language
+
+ Returns:
+ Text with translated keywords
+ """
+ # Basic keyword translation
+ thai_to_english = {
+ 'เสียงพิณ': 'pin sound',
+ 'หมอลำ': 'molam',
+ 'เพลิน': 'contemplative',
+ 'ซิ่ง': 'fast',
+ 'กลอน': 'poetic',
+ 'ตัด': 'narrative',
+ 'ปึน': 'storytelling',
+ 'เศร้า': 'sad',
+ 'สนุก': 'fun',
+ 'โรแมนติก': 'romantic',
+ 'ช้า': 'slow',
+ 'เร็ว': 'fast',
+ 'ปานกลาง': 'medium',
+ 'ดนตรี': 'music',
+ 'เพลง': 'song',
+ 'จังหวะ': 'rhythm',
+ 'ทำนอง': 'melody',
+ }
+
+ if target_lang == 'en':
+ # Replace Thai terms with English
+ for thai, english in thai_to_english.items():
+ text = text.replace(thai, english)
+
+ return text
+
+
+def setup_directories(base_dir: str = None) -> Dict[str, Path]:
+ """
+ Setup necessary directories for the project
+
+ Args:
+ base_dir: Base directory (uses default if None)
+
+ Returns:
+ Dictionary of directory paths
+ """
+ if base_dir is None:
+ base_dir = Path(__file__).parent.parent
+ else:
+ base_dir = Path(base_dir)
+
+ directories = {
+ 'base': base_dir,
+ 'audio': base_dir / 'audio',
+ 'models': base_dir / 'models',
+ 'logs': base_dir / 'logs',
+ 'temp': base_dir / 'temp',
+ 'cache': base_dir / 'cache',
+ 'data': base_dir / 'data',
+ 'output': base_dir / 'output',
+ }
+
+ # Create directories
+ for dir_path in directories.values():
+ if isinstance(dir_path, Path):
+ dir_path.mkdir(parents=True, exist_ok=True)
+
+ return directories
+
+
+def get_file_hash(file_path: Union[str, Path]) -> str:
+ """
+ Get MD5 hash of a file
+
+ Args:
+ file_path: Path to file
+
+ Returns:
+ MD5 hash string
+ """
+ import hashlib
+
+ hash_md5 = hashlib.md5()
+ with open(file_path, "rb") as f:
+ for chunk in iter(lambda: f.read(4096), b""):
+ hash_md5.update(chunk)
+ return hash_md5.hexdigest()
+
+
+def format_duration(seconds: float) -> str:
+ """
+ Format duration in seconds to human readable string
+
+ Args:
+ seconds: Duration in seconds
+
+ Returns:
+ Formatted duration string
+ """
+ if seconds < 60:
+ return f"{seconds:.1f}s"
+ elif seconds < 3600:
+ minutes = seconds / 60
+ return f"{minutes:.1f}m"
+ else:
+ hours = seconds / 3600
+ return f"{hours:.1f}h"
+
+
+def format_file_size(bytes_size: int) -> str:
+ """
+ Format file size in bytes to human readable string
+
+ Args:
+ bytes_size: Size in bytes
+
+ Returns:
+ Formatted size string
+ """
+ for unit in ['B', 'KB', 'MB', 'GB']:
+ if bytes_size < 1024.0:
+ return f"{bytes_size:.1f} {unit}"
+ bytes_size /= 1024.0
+ return f"{bytes_size:.1f} TB"
+
+
+# Make AudioUtils available as a direct import
+# This maintains backward compatibility for existing imports
+AudioUtils = AudioUtils
+TextProcessor = TextProcessor
\ No newline at end of file
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