--- license: apache-2.0 task_categories: - audio-to-audio tags: - audio-super-resolution --- # LJSpeech-1.1 High-Resolution Dataset (48,000 Hz) This dataset was created using the method described in [HiFi-SR: A Unified Generative Transformer-Convolutional Adversarial Network for High-Fidelity Speech Super-Resolution](https://huggingface.co./papers/2501.10045). The LJSpeech-1.1 dataset, widely recognized for its utility in text-to-speech (TTS) and other speech processing tasks, has now been enhanced through a cutting-edge speech super-resolution algorithm. The original dataset, which featured a sampling rate of 22,050 Hz, has been upscaled to 48,000 Hz using [**ClearerVoice-Studio**](https://github.com/modelscope/ClearerVoice-Studio), providing a high-fidelity version suitable for advanced audio processing tasks. **Key Features** - High-Resolution Audio: The dataset now offers audio files at a sampling rate of 48,000 Hz, delivering enhanced perceptual quality with richer high-frequency details. - Original Content Integrity: The original linguistic content and annotation structure remain unchanged, ensuring compatibility with existing workflows. - Broader Application Scope: Suitable for professional-grade audio synthesis, TTS systems, and other high-quality audio applications. - Open Source: Freely available for academic and research purposes, fostering innovation in the speech and audio domains. **Original Dataset** - Source: The original LJSpeech-1.1 dataset contains 13,100 audio clips of a single female speaker reading passages from public domain books. - Duration: Approximately 24 hours of speech data. - Annotations: Each audio clip is paired with a corresponding text transcript. **Super-Resolution Processing** The original 22,050 Hz audio recordings were processed using a state-of-the-art MossFormer2-based speech super-resolution model. This model employs: - Advanced Neural Architectures: A combination of transformer-based sequence modeling and convolutional networks. - Perceptual Optimization: Loss functions designed to preserve the naturalness and clarity of speech. - High-Frequency Reconstruction: Algorithms specifically tuned to recover lost high-frequency components, ensuring smooth and artifact-free enhancement. **Output Format** - Sampling Rate: 48,000 Hz - Audio Format: WAV - Bit Depth: 16-bit - Channel Configuration: Mono **Use Cases** 1. Text-to-Speech (TTS) Synthesis ○ Train high-fidelity TTS systems capable of generating human-like speech. ○ Enable expressive and emotionally nuanced TTS outputs. 2. Speech Super-Resolution Benchmarking ○ Serve as a reference dataset for evaluating speech super-resolution algorithms. ○ Provide a standardized benchmark for perceptual quality metrics. 3. Audio Enhancement and Restoration ○ Restore low-resolution or degraded speech signals for professional applications. ○ Create high-quality voiceovers and narration for multimedia projects. **File Structure** The dataset retains the original LJSpeech-1.1 structure, ensuring ease of use: ```sh LJSpeech-1.1-48kHz/ ├── metadata.csv # Text transcriptions and audio file mappings ├── wavs/ # Directory containing 48,000 Hz WAV files └── LICENSE.txt # License information ``` **Licensing** The LJSpeech-1.1 High-Resolution Dataset is released under the same open license as the original LJSpeech-1.1 dataset (https://keithito.com/LJ-Speech-Dataset/). Users are free to use, modify, and share the dataset for academic and non-commercial purposes, provided proper attribution is given.