ccMixter
The premier open-source repository for Creative Commons-licensed stems, acappellas, and collaborative remixes.

The all-in-one, SOTA open-source toolkit for high-performance speech recognition and synthesis.
PaddleSpeech is a comprehensive, open-source speech library built on the PaddlePaddle deep learning framework. As of 2026, it remains a dominant force in the Mandarin and English speech processing landscape, offering a unified architecture for Automatic Speech Recognition (ASR), Text-to-Speech (TTS), Speaker Verification, and Audio Classification. Its technical architecture is designed for modularity, allowing developers to switch between state-of-the-art models like Conformer, U2, and FastSpeech2 with minimal configuration changes. PaddleSpeech excels in low-latency streaming scenarios and offers a complete toolchain from model training and fine-tuning to deployment on edge devices or cloud servers. Its market position is unique, bridging the gap between academic research and industrial-grade production, particularly favored by enterprises seeking to avoid vendor lock-in while maintaining performance comparable to commercial APIs like OpenAI Whisper or Google Speech-to-Text. With integrated support for model compression and quantization, it is highly optimized for deployment on diverse hardware environments including ARM, NVIDIA GPUs, and specialized AI accelerators.
Supports real-time streaming inference using chunk-based processing for low-latency feedback.
The premier open-source repository for Creative Commons-licensed stems, acappellas, and collaborative remixes.
Transform raw audio into a multi-channel content engine using high-fidelity AI transcription and semantic repurposing.
Advanced real-time voice morphing and audio manipulation for professional creators and streamers.
Professional-grade AI audio mastering for studio-quality sound in seconds.
Verified feedback from the global deployment network.
Post queries, share implementation strategies, and help other users.
Implements ECAPA-TDNN and ResNet-based models for high-accuracy voice fingerprinting.
A dedicated NLP module that adds commas, periods, and question marks to raw ASR output.
Pre-trained models for Mandarin, English, and multi-dialect support for Chinese regions.
Supports INT8 and FP16 quantization via PaddleSlim for mobile deployment.
Optimized models for 'Wake Word' detection with extremely low false-alarm rates.
Identifies environmental sounds, music genres, or emotional states in audio files.
Providing accessibility and clarity for live broadcasts with minimal delay.
Registry Updated:2/7/2026
Render subtitles on top of the video feed.
Creating a consistent brand voice for internal training videos and customer service.
Analyzing thousands of hours of customer calls for compliance and quality.