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OpenBMB/VoxCPM

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29,985 stars·3,396 forks·Python·Apache-2.0·59 views

VoxCPM

VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice cloning tool and a synthetic voice designer, capable of generating natural speech across global languages and regional dialects using a GPU-accelerated audio generator.

The project features a speech model fine-tuning framework that supports both full parameter updates and low-rank adaptation for customizing voice characteristics. It enables high-fidelity voice cloning from reference audio, including cross-lingual voice transfer and acoustic environment mimicry, as well as the creation of unique vocal identities through text-based voice design.

The system provides broad capabilities for speech generation, including context-aware prosody, non-verbal cue insertion, and multi-speaker dialogue. It includes professional audio processing utilities for denoising and upsampling reference clips, as well as a high-throughput API server with streaming output and an OpenAI-compatible interface.

The software supports deployment across various hardware backends, including CUDA, MPS, and CPU, and can be deployed via containers.

Features

  • Multilingual Speech Models - Synthesizes natural speech across global languages and regional dialects without requiring language tags.
  • Speech Synthesis - Functions as a high-performance speech synthesis server that converts text to audio via an HTTP interface.
  • Text-to-Speech - Synthesizes natural-sounding speech across global languages and regional dialects using local generative inference.
  • Adapter Fine-Tuning - Uses Low-Rank Adaptation (LoRA) to fine-tune specific model layers for efficient voice cloning.
  • Autoregressive Audio Diffusion - Uses a hierarchical diffusion-based autoregressive architecture to generate high-fidelity continuous speech representations.
  • Zero-Shot Voice Cloning - Analyzes timbre and prosody from short reference audio samples to enable zero-shot voice cloning.
  • Generative Audio Engines - Implements a GPU-accelerated inference engine optimized for CUDA and MPS to produce studio-quality audio.
  • Speech Model Fine-Tuning - Provides a toolkit for adapting speech models using full parameter updates or LoRA adapters for custom voice characteristics.
  • High-Throughput Model Serving - Ships a dedicated high-throughput inference engine with asynchronous APIs and concurrent request support.
  • Hardware-Agnostic Inference Layers - Implements a standardized runtime format that enables model execution across CUDA, MPS, and CPU backends.
  • Model Fine-Tuning - Provides a framework for adapting speech models to specific speakers or languages via full fine-tuning and LoRA.
  • Speech Synthesis Gateways - Hosts a high-throughput HTTP server that acts as a gateway for streaming audio generation to external clients.
  • Conversational Audio Streams - Implements a network endpoint that streams generated speech as an MP3 byte stream for real-time interaction.
  • Local Speech Synthesis - Implements a high-performance text-to-speech engine running on local CUDA, MPS, or CPU backends.
  • Synthetic Voice Generators - Allows the creation of unique synthetic vocal identities by describing attributes like age, gender, and emotion in plain text.
  • Text-Based Voice Design - Allows the creation of unique synthetic vocal identities based on natural-language descriptions of age, gender, and pitch.
  • Voice Cloning - Replicates specific human vocal identities using reference audio samples and high-fidelity synthesis.
  • Voice Profile Managers - Offers a programmatic interface to register and organize custom voice profiles for audio customization.
  • OpenAI-Compatible APIs - Provides a local server implementation that accepts OpenAI-standard speech requests for broad ecosystem compatibility.
  • Audio Noise Cancellation - Provides utilities to remove background noise from reference audio clips to improve the quality of voice cloning.
  • Audio Processing - Offers professional audio utilities for denoising reference clips and upsampling low-resolution samples to studio quality.
  • Audio Transcription - Includes a speech-to-text utility that converts reference audio clips into text to streamline cloning.
  • Sample-Rate Conditioned Decoding - Produces high-resolution studio audio by conditioning the output decoder on specific target sample rates.
  • Full Parameter Fine-Tuning - Provides workflows for updating all model parameters to achieve maximum voice synthesis performance.
  • Continuous Batching Strategies - Optimizes GPU memory and throughput using continuous batching and attention paging for concurrent speech generation.
  • Cross-Lingual Adaptation - Enables speech generation for unsupported languages through specialized fine-tuning on target-language datasets.
  • Hardware-Agnostic Deployment - Implements a standardized model format that enables speech synthesis across diverse CPU and GPU backends.
  • LoRA Adapter Loaders - Provides a mechanism to load optional Low-Rank Adaptation weights to refine the output of the speech generation model.
  • Multi-Speaker Synthesis - Produces audio for multiple speakers using tagged scripts to assign distinct voices to different parts.
  • Supervised Fine-Tuning - Adapts models to specific speakers or languages using labeled datasets and supervised fine-tuning.
  • Acoustic Style Controls - Modifies emotion, pace, and delivery of audio using text-based control tags.
  • Phonetic Pronunciation Overrides - Allows precise control over pronunciation by overriding standard text with phoneme inputs or pinyin.
  • Prosody Controls - Infers prosody and expressiveness directly from text to produce natural, context-matched speech delivery.
  • Tokenizer-Free Processing - Processes multilingual text input directly without relying on predefined vocabulary tokens to maintain linguistic flexibility.
  • Acoustic Environment Replication - Reproduces both the speaker's identity and the specific acoustic environment of the source audio.
  • Cross-Lingual Voice Transfer - Clones a speaker's voice from a reference audio file and applies it to a different target language.
  • Prosody and Style Control - Enables duplication of a speaker's timbre while allowing precise adjustment of speed and emotion via text.
  • Speech Dialects - Provides the ability to replicate regional accents and dialects across multiple languages during speech synthesis.
  • Inference Scaling Services - Features a system for distributing concurrent synthesis requests across a pool of GPUs to maximize throughput.
  • Multi-GPU Deployment - Includes a distribution system that spreads model weights and computation across multiple GPUs for larger loads.
  • Emotional Modulation - Implements emotional modulation to adjust the tone and intensity of synthetic speech to match target emotions.
  • Audio Super-Resolution - Produces high-resolution audio from lower-sample-rate references using built-in upsampling.
  • Chunked Audio Streaming - Streams audio waveform chunks sequentially so playback can begin before the full synthesis completes.
  • Generative Audio Chunking - Yields audio waveform chunks sequentially during generation to allow playback to begin before the full sequence is complete.
  • Asynchronous Task Queues - Manages concurrent synthesis tasks using an asynchronous server that handles task priority and GPU memory allocation.
  • Non-Verbal Audio Cues - Adds realistic non-verbal cues like laughs and sighs into generated audio using specific text tags.
  • TTS API Endpoints - Provides a standard request-response endpoint for integrating text-to-speech functionality into external applications.
  • AI and Agents - Listed in the “AI and Agents” section of the Awesome Python awesome list.
  • Speech Processing - Multimodal speech and language model.
  • Speech Synthesis - Multimodal speech and language model.

Star history

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How this analysis was created: This summary and feature list are AI-generated from collected project material and can contain mistakes. Stars, license and language are imported from GitHub. Inclusion does not mean that we have tested or audited this project. Check the source documentation for any feature you depend on. Learn more on our About page.

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Frequently asked questions

What does openbmb/voxcpm do?

VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice cloning tool and a synthetic voice designer, capable of generating natural speech across global languages and regional dialects using a GPU-accelerated audio generator.

What are the main features of openbmb/voxcpm?

The main features of openbmb/voxcpm are: Multilingual Speech Models, Speech Synthesis, Text-to-Speech, Adapter Fine-Tuning, Autoregressive Audio Diffusion, Zero-Shot Voice Cloning, Generative Audio Engines, Speech Model Fine-Tuning.

Which projects share features with openbmb/voxcpm?

Projects with overlapping indexed features include: livekit/agents — This project is a framework for developing multimodal AI agents that function as programmable participants in… boson-ai/higgs-audio — Higgs-audio is a generative text-to-speech engine that transforms text into natural conversational speech using large… k2-fsa/sherpa-onnx — Sherpa-ONNX is an ONNX-based speech processing toolkit that provides a local speech recognition engine, an on-device… livekit/livekit — LiveKit is a comprehensive framework for building and orchestrating real-time, multimodal AI agents that interact with… qwenlm/qwen3-tts — Qwen3-TTS is a large language model text-to-speech engine designed to convert written text into natural-sounding human… neonbjb/tortoise-tts — Tortoise-tts is a neural text-to-speech engine and voice cloning toolkit designed for high-quality audio generation.…