Amphion Toolkit

repository·main·Indexed 27 days ago

https://github.com/open-mmlab/amphion

An open-source toolkit for Audio, Music, and Speech Generation designed for reproducible research and model architecture visualization. It includes implementations for FAcodec (neural audio codec), DiffComoSVC, and MultipleContentsSVC for singing voice conversion, as well as tools for objective evaluation metrics (F0, energy, intelligibility, and speaker similarity) and support for various academic datasets like VCTK.

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What's inside Amphion

  1. Overview of Amphion Toolkit

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    Amphion is an open-source toolkit designed for Audio, Music, and Speech Generation research and development. It aims to provide a platform for converting various inputs into audio and supports reproducible research through model visualizations.

    Key supported generation tasks include:

    • TTS: Text to Speech
    • SVS: Singing Voice Synthesis
    • VC: Voice Conversion
    • AC: Accent Conversion
    • SVC: Singing Voice Conversion
    • TTA: Text to Audio
    • TTM: Text to Music (currently under development)

    The toolkit also provides vocoders for high-quality audio signal production and evaluation metrics for consistent performance measurement in generation tasks.

  2. Overview of Amphion Model Releases

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    Amphion is an open-source toolkit for audio, music, and speech generation. The project has released several key models and datasets:

    • Vevo2: A unified framework for speech and singing voice generation (TTS, SVS, VC, SVC, Singing Style Conversion, etc.).
    • DualCodec: A low-frame-rate (12.5Hz or 25Hz) semantically-enhanced Neural Audio Codec for efficient speech generation.
    • Vevo1.5: An extension of Vevo focusing on controllable generation for speech and singing.
    • Metis: A foundation model for unified speech generation (zero-shot TTS, VC, target speaker extraction, speech enhancement, and lip-to-speech).
    • Vevo: A zero-shot voice imitation framework with controllable timbre and style.
    • MaskGCT: A fully non-autoregressive TTS model that eliminates the need for explicit text-speech alignment.
    • NaturalSpeech3 FACodec: Supported codec with pretrained checkpoints.

    Datasets & Tools:

    • Emilia-Large: A massive dataset (200,000+ hours) combining the original Emilia and Emilia-YODAS datasets.
    • Emilia-Pipe: A preprocessing pipeline for transforming in-the-wild speech data into high-quality training data.
    • SingVisio: A visualization tool for singing voice research.
  3. Overview of GAN-based Vocoder Recipe

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    The Amphion GAN-based Vocoder recipe allows you to train and use various GAN-based vocoder architectures. The process follows four main stages: Data Preparation, Feature Extraction, Training, and Inference.

    Supported Generators:

    • MelGAN
    • HiFi-GAN
    • NSF-HiFiGAN
    • BigVGAN
    • APNet

    Supported Discriminators:

    • Multi-Scale Discriminator (msd)
    • Multi-Period Discriminator (mpd)
    • Multi-Resolution Discriminator
    • Multi-Scale Short-Time Fourier Transform Discriminator (msstftd)
    • Multi-Scale Constant-Q Transform Discriminator (mssbcqtd)
  4. Overview of Emilia and Emilia-Pipe

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    Emilia is a large-scale, multilingual speech generation dataset. Emilia-Large contains over 215k hours of speech data across English, Chinese, German, French, Japanese, and Korean.

    Emilia-Pipe is an open-source preprocessing pipeline designed to transform raw, in-the-wild speech data into high-quality, annotated training data for speech generation. The pipeline follows these steps:

    1. Standardization: Audio normalization.
    2. Source Separation: Separating speech from background music (BGM).
    3. Speaker Diarization: Extracting single-speaker speech.
    4. Fine-grained Segmentation (VAD): Creating 3-30s single-speaker segments.
    5. ASR: Generating transcriptions.
    6. Filtering: Final dataset refinement.
  5. Overview of MultipleContentsSVC

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    MultipleContentsSVC is an implementation for Singing Voice Conversion (SVC) that leverages diverse semantic-based audio pretrained models (Whisper and ContentVec) to extract content features. The acoustic model uses DiffWaveNetSVC (a Bidirectional Non-Causal Dilated CNN), and the vocoder uses the BigVGAN architecture.

    To use this recipe, all commands must be executed from the Amphion root directory.

  6. Overview of SingVisio Visual Analytics System

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    SingVisio is a visual analytics system designed to analyze the diffusion models used in Singing Voice Conversion (SVC). It consists of a web-based front-end (built with D3.js) and a back-end generative model.

    Core Components

    • Web Interface: The front-end code is located in ../../../visualization/SingVisio/webpage/.
    • Generative Model: Uses MultipleContentsSVC, a diffusion-based model. The recipe for this model can be found in the Amphion repository under ../../svc/MultipleContentsSVC/.

    Online Experience

    You can experience the online system via OpenXLab.

  7. Explore Amphion Voice Conversion (VC) models

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    Amphion provides several models for voice conversion:

    • Vevo: Zero-shot imitation. Use Vevo-Timbre for style-preserved conversion and Vevo-Voice for style-converted conversion.
    • FACodec: Decomposes speech into content, prosody, and timbre subspaces for zero-shot VC.
    • Noro: A noise-robust zero-shot VC system using dual-branch reference encoding and noise-agnostic contrastive speaker loss.
  8. Explore Amphion Text-to-Speech (TTS) models

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    Amphion supports several state-of-the-art TTS architectures:

    • FastSpeech2: Non-autoregressive Transformer-based.
    • VITS: End-to-end with conditional VAE and adversarial learning.
    • VALL-E: Zero-shot neural codec language model.
    • NaturalSpeech2: Latent diffusion model.
    • Jets: Jointly trained FastSpeech2 and HiFi-GAN.
    • MaskGCT: Fully non-autoregressive without explicit alignment.
    • Vevo-TTS: Zero-shot with controllable timbre and style (autoregressive + flow-matching).
    • DualCodec-VALLE: VALL-E model using 12.5Hz DualCodec tokens.
  9. Explore Amphion Vocoders

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    Amphion supports various neural vocoders:

    • GAN-based: MelGAN, HiFi-GAN, NSF-HiFiGAN, BigVGAN, APNet.
    • Flow-based: WaveGlow.
    • Diffusion-based: Diffwave.
    • Auto-regressive: WaveNet, WaveRNN.
    • Enhancement: Supports the Multi-Scale Constant-Q Transform Discriminator to enhance GAN-based vocoders during training without affecting inference.
  10. Supported Model Architectures for Amphion SVC

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    Amphion's SVC pipeline disentangles speaker-agnostic representations from source audio and injects speaker information using an acoustic decoder and a waveform synthesizer (vocoder). Supported components include:

    Speaker-agnostic Representations

    • Content Features: Sourced from WeNet, Whisper, and ContentVec.
    • Prosody Features: F0 and energy.

    Speaker Embeddings

    • Speaker Look-Up Table.
    • Reference Encoder (currently under development): Enables zero-shot SVC.

    Acoustic Decoders

    • Diffusion-based models:
      • DiffWaveNetSVC: Uses a Bidirectional Non-Causal Dilated CNN encoder (similar to WaveNet, DiffWave, and DiffSVC).
      • DiffComoSVC (under development): Based on the Consistency Model framework to accelerate inference.
    • Transformer-based models:
      • TransformerSVC: Uses an encoder-only and non-autoregressive Transformer architecture.
    • VAE- and Flow-based models:
      • VitsSVC: A VITS-like model where textual input is replaced by content features (similar to so-vits-svc).

    Waveform Synthesizers (Vocoders)

  11. Explore Amphion Neural Audio Codecs

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    Amphion implements high-efficiency audio codecs:

    • DualCodec: A low-frame-rate (12.5Hz or 25Hz) semantically-enhanced codec designed for efficient speech generation.
    • FACodec: Decomposes speech into subspaces (content, prosody, timbre) for attribute manipulation.
  12. Supported Model Architectures for Voice Conversion (VC)

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    Amphion currently supports the NORO architecture for noise-robust Voice Conversion. Key features of this architecture include:

    1. Noise-Robust Voice Conversion: Uses a dual-branch reference encoding module and noise-agnostic contrastive speaker loss to ensure high-quality conversion even in noisy environments.
    2. One-shot Voice Conversion: Enables timbre conversion using only a single reference speech sample.
    3. Speaker Representation Learning: Leverages the reference encoder as a self-supervised speaker encoder.