Rhasspy Larynx
High-quality, privacy-first neural text-to-speech for local edge computing.
Ultra-fast, non-autoregressive neural speech synthesis with explicit prosody control.
FastSpeech 2 is a state-of-the-art non-autoregressive text-to-speech (TTS) architecture developed by Microsoft Research. Unlike its predecessor, FastSpeech, which relied on a teacher-student distillation process, FastSpeech 2 utilizes a simplified training pipeline by directly predicting speech parameters from the ground truth. The model architecture is built on a Feed-Forward Transformer that enables parallelized mel-spectrogram generation, significantly reducing inference latency compared to autoregressive models like Tacotron 2. A critical technical innovation in FastSpeech 2 is the introduction of Variance Adapters, which explicitly predict duration, pitch, and energy. This allows for fine-grained control over prosody and addresses the 'one-to-many' mapping problem in TTS, where the same text can be spoken in various ways. In the 2026 market, FastSpeech 2 remains a foundational architecture for edge computing and real-time voice applications due to its computational efficiency and robust alignment stability. It is widely implemented via frameworks like ESPnet, Fairseq, and TensorSpeech, making it the industry standard for developers requiring high-fidelity voice output without the high overhead of diffusion-based or massive autoregressive models.
Generates mel-spectrograms in parallel rather than sequentially.
High-quality, privacy-first neural text-to-speech for local edge computing.
A high-speed, fully convolutional neural architecture for multi-speaker text-to-speech synthesis.
Real-time neural text-to-speech architecture for massive-scale multi-speaker synthesis.
A Multilingual Single-Speaker Speech Corpus for High-Fidelity Text-to-Speech Synthesis.
Verified feedback from the global deployment network.
Post queries, share implementation strategies, and help other users.
A module that explicitly predicts duration, pitch, and energy for each phoneme.
Uses ground-truth targets directly instead of distilling from an autoregressive model.
Maps phoneme sequences to mel-spectrogram frames based on duration prediction.
Allows manual adjustment of predicted F0 and energy values during inference.
Uses hard alignment instead of soft attention mechanisms.
Can be conditioned on speaker embeddings for zero-shot or few-shot voice cloning.
Latency in AI responses causes unnatural conversational pauses.
Registry Updated:2/7/2026
High cost of human narrators for long-form content.
Communication devices lack personalized, natural-sounding voices.