Collaborating Authors

Speech Synthesis

VALL-E -- The Future of Text to Speech?


In this article, we will dive deep into a new and exciting text-to-speech model developed by Microsoft Research, called VALL-E. The paper presenting the work has been released on Jan. 5, 2023, and since then has been gaining much attention online. It is worth noting that as of writing this article, no pre-trained model has been released and the only option currently to battle-test this model is to train it by yourself. Nevertheless, the idea presented in this paper is novel and interesting and worth digging into, regardless of whether I can immediately clone my voice with it or not. The technology of text-to-speech is not new and has been around since the "Voder" -- the first electronic voice synthesizer from Bell Labs in 1939 which required manual operation.

Deep Voice 2: Multi-Speaker Neural Text-to-Speech

Neural Information Processing Systems

We introduce a technique for augmenting neural text-to-speech (TTS) with lowdimensional trainable speaker embeddings to generate different voices from a single model. As a starting point, we show improvements over the two state-ofthe-art approaches for single-speaker neural TTS: Deep Voice 1 and Tacotron. We introduce Deep Voice 2, which is based on a similar pipeline with Deep Voice 1, but constructed with higher performance building blocks and demonstrates a significant audio quality improvement over Deep Voice 1. We improve Tacotron by introducing a post-processing neural vocoder, and demonstrate a significant audio quality improvement. We then demonstrate our technique for multi-speaker speech synthesis for both Deep Voice 2 and Tacotron on two multi-speaker TTS datasets. We show that a single neural TTS system can learn hundreds of unique voices from less than half an hour of data per speaker, while achieving high audio quality synthesis and preserving the speaker identities almost perfectly.

Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis

Neural Information Processing Systems

We describe a neural network-based system for text-to-speech (TTS) synthesis that is able to generate speech audio in the voice of different speakers, including those unseen during training. Our system consists of three independently trained components: (1) a speaker encoder network, trained on a speaker verification task using an independent dataset of noisy speech without transcripts from thousands of speakers, to generate a fixed-dimensional embedding vector from only seconds of reference speech from a target speaker; (2) a sequence-to-sequence synthesis network based on Tacotron 2 that generates a mel spectrogram from text, conditioned on the speaker embedding; (3) an auto-regressive WaveNet-based vocoder network that converts the mel spectrogram into time domain waveform samples. We demonstrate that the proposed model is able to transfer the knowledge of speaker variability learned by the discriminatively-trained speaker encoder to the multispeaker TTS task, and is able to synthesize natural speech from speakers unseen during training. We quantify the importance of training the speaker encoder on a large and diverse speaker set in order to obtain the best generalization performance. Finally, we show that randomly sampled speaker embeddings can be used to synthesize speech in the voice of novel speakers dissimilar from those used in training, indicating that the model has learned a high quality speaker representation.

FastSpeech: Fast, Robust and Controllable Text to Speech

Neural Information Processing Systems

Neural network based end-to-end text to speech (TTS) has significantly improved the quality of synthesized speech. Prominent methods (e.g., Tacotron 2) usually first generate mel-spectrogram from text, and then synthesize speech from the mel-spectrogram using vocoder such as WaveNet. Compared with traditional concatenative and statistical parametric approaches, neural network based endto-end models suffer from slow inference speed, and the synthesized speech is usually not robust (i.e., some words are skipped or repeated) and lack of controllability (voice speed or prosody control). In this work, we propose a novel feed-forward network based on Transformer to generate mel-spectrogram in parallel for TTS. Specifically, we extract attention alignments from an encoder-decoder based teacher model for phoneme duration prediction, which is used by a length regulator to expand the source phoneme sequence to match the length of the target mel-spectrogram sequence for parallel mel-spectrogram generation. Experiments on the LJSpeech dataset show that our parallel model matches autoregressive models in terms of speech quality, nearly eliminates the problem of word skipping and repeating in particularly hard cases, and can adjust voice speed smoothly. Most importantly, compared with autoregressive Transformer TTS, our model speeds up mel-spectrogram generation by 270x and the end-to-end speech synthesis by 38x. Therefore, we call our model FastSpeech.

Supplementary Material of HiFi-GAN: Generative Adversarial Networks for Efficient and High Fidelity Speech Synthesis

Neural Information Processing Systems

Details of the Model Architecture The detailed architecture of the generator and MPD is depicted in Figure 4. The configuration of three variants of the generator is listed in Table 5. In the ResBlock of V1 and V2, 2 convolution layers and 1 residual connection are stacked 3 times. In the Resblock of V3, 1 convolution layer and 1 residual connection are stacked 2 times. Therefore, V3 consists of a much smaller number of layers than V1 and V2. Periodic signal discrimination experiments We conducted additional experiments similar to training a discriminator using a simple dataset to verify the ability of MPD to discriminate periodic signals.

Appendix A A proper scoring rule for speech synthesis

Neural Information Processing Systems

A loss function or scoring rule L(q, x) measures how well a model distribution q fits data x drawn from a distribution p. Such a scoring rule is called proper if its expectation is minimized when q = p. If the minimum is also unique, the scoring rule is called strictly proper. In the large data limit, a strictly proper scoring rule can uniquely identify the distribution p, which means that it can be used as the basis of a statistically consistent learning method. This includes the special cases of L1 and L2 distance, the latter of which they show leads to a strictly proper scoring rule.

Supplementary Material of Glow-TTS: A Generative Flow for Text-to-Speech via Monotonic Alignment Search

Neural Information Processing Systems

Details of the Model Architecture The detailed encoder architecture is depicted in Figure 7. Some implementation details that we use in the decoder, and the decoder architecture are depicted in Figure 8. We design the grouped 1x1 convolutions to be able to mix channels. For each group, the same number of channels are extracted from one half of the feature map separated by coupling layers and the other half, respectively. Figure 8c shows an example.

Glow-TTS: A Generative Flow for Text-to-Speech via Monotonic Alignment Search

Neural Information Processing Systems

Recently, text-to-speech (TTS) models such as FastSpeech and ParaNet have been proposed to generate mel-spectrograms from text in parallel. Despite the advantage, the parallel TTS models cannot be trained without guidance from autoregressive TTS models as their external aligners. In this work, we propose Glow-TTS, a flow-based generative model for parallel TTS that does not require any external aligner. By combining the properties of flows and dynamic programming, the proposed model searches for the most probable monotonic alignment between text and the latent representation of speech on its own. We demonstrate that enforcing hard monotonic alignments enables robust TTS, which generalizes to long utterances, and employing generative flows enables fast, diverse, and controllable speech synthesis.

Modular Meta-Learning with Shrinkage

Neural Information Processing Systems

Many real-world problems, including multi-speaker text-to-speech synthesis, can greatly benefit from the ability to meta-learn large models with only a few taskspecific components. Updating only these task-specific modules then allows the model to be adapted to low-data tasks for as many steps as necessary without risking overfitting. Unfortunately, existing meta-learning methods either do not scale to long adaptation or else rely on handcrafted task-specific architectures. Here, we propose a meta-learning approach that obviates the need for this often sub-optimal hand-selection. In particular, we develop general techniques based on Bayesian shrinkage to automatically discover and learn both task-specific and general reusable modules. Empirically, we demonstrate that our method discovers a small set of meaningful task-specific modules and outperforms existing metalearning approaches in domains like few-shot text-to-speech that have little task data and long adaptation horizons. We also show that existing meta-learning methods including MAML, iMAML, and Reptile emerge as special cases of our method.

PortaSpeech: Portable and High-Quality Generative Text-to-Speech

Neural Information Processing Systems

Non-autoregressive text-to-speech (NAR-TTS) models such as FastSpeech 2 [24] and Glow-TTS [8] can synthesize high-quality speech from the given text in parallel. After analyzing two kinds of generative NAR-TTS models (VAE and normalizing flow), we find that: VAE is good at capturing the long-range semantics features (e.g., prosody) even with small model size but suffers from blurry and unnatural results; and normalizing flow is good at reconstructing the frequency bin-wise details but performs poorly when the number of model parameters is limited. Inspired by these observations, to generate diverse speech with natural details and rich prosody using a lightweight architecture, we propose PortaSpeech, a portable and high-quality generative text-to-speech model. Specifically, 1) to model both the prosody and mel-spectrogram details accurately, we adopt a lightweight VAE with an enhanced prior followed by a flow-based post-net with strong conditional inputs as the main architecture.