
Retrieval-based Voice Conversion WebUI
Easily train a good VC model with voice data in <= 10 mins!

High-quality, low-complexity neural vocoder combining DSP and Deep Learning for real-time speech synthesis.

LPCNet is a pioneering hybrid neural vocoder that integrates traditional Digital Signal Processing (DSP) techniques, specifically Linear Predictive Coding (LPC), with deep recurrent neural networks (RNN). Developed primarily by Jean-Marc Valin at Mozilla, it represents a significant leap in audio synthesis efficiency, enabling high-quality speech generation at computational loads significantly lower than pure-neural models like WaveNet. By using the LPC coefficients to handle the spectral envelope, the neural network only needs to model the residual excitation signal, which is much easier to learn and requires fewer parameters. As of 2026, LPCNet has become a foundational architecture for low-bitrate speech codecs and real-time Text-to-Speech (TTS) applications on edge devices. It utilizes sparse GRU (Gated Recurrent Unit) layers and 8-bit quantization to achieve real-time performance on high-end mobile CPUs without requiring dedicated GPU acceleration. This makes it ideal for privacy-focused, on-device voice synthesis and low-latency communication protocols where bandwidth and power are constrained.
LPCNet is a pioneering hybrid neural vocoder that integrates traditional Digital Signal Processing (DSP) techniques, specifically Linear Predictive Coding (LPC), with deep recurrent neural networks (RNN).
Explore all tools that specialize in synthesize speech. This domain focus ensures LPCNet delivers optimized results for this specific requirement.
Explore all tools that specialize in neural vocoding. This domain focus ensures LPCNet delivers optimized results for this specific requirement.
Combines a linear prediction filter with a gated recurrent unit to reduce the complexity of the neural synthesis task.
Uses structured sparsity in the GRU layers to skip redundant computations during inference.
Processes coarse-grained spectral features at a lower rate than the sample-level excitation.
Outputs audio in 8-bit u-law format internally to simplify the probability distribution modeling.
Adjusts neural processing based on the fundamental frequency of the input speech.
Predicts missing audio frames using the neural network's stateful memory.
Hand-rolled intrinsics for Intel AVX2 and ARM NEON architectures.
Clone the official LPCNet repository from GitHub.
Install dependencies including a C compiler (GCC/Clang) and Python 3.x.
Install Keras and TensorFlow for the training environment.
Prepare the speech dataset (e.g., LJSpeech or internal high-quality PCM files).
Run the feature extraction script to generate pitch and spectral data.
Configure the model hyperparameters in the training script (GRU size, sparsity).
Execute the training process (typically requires a GPU for efficient convergence).
Convert the trained Keras model weights into C headers using the provided conversion scripts.
Compile the C inference engine with AVX2 or NEON optimization flags enabled.
Integrate the resulting library into your application for real-time synthesis.
All Set
Ready to go
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"Highly regarded in the research community for its efficiency/quality ratio. Users praise its ability to run on edge hardware where other models fail."
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Easily train a good VC model with voice data in <= 10 mins!

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