E ND - TO -E ND A DVERSARIAL T EXT - TO -S PEECH
A BSTRACT
Modern text-to-speech synthesis pipelines typically involve multiple processing stages, each of which is designed or learnt independently from the rest.
In this work, we take on the challenging task of learning to synthesise speech from normalised text or phonemes in an end-to-end manner, resulting in models which operate directly on character or phoneme input sequences and produce raw speech audio outputs.
Our proposed generator is feed-forward and thus efﬁcient for both training and inference, using a differentiable alignment scheme based on token length prediction.
It learns to produce high ﬁdelity audio through a combination of adversarial feedback and prediction losses constraining the generated audio to roughly match the ground truth in terms of its total duration and mel-spectrogram.
To allow the model to capture temporal variation in the generated audio, we employ soft dynamic time warping in the spectrogram-based prediction loss.
The resulting model achieves a mean opinion score exceeding 4 on a 5 point scale, which is comparable to the state-of-the-art models relying on multi-stage training and additional supervision.
I NTRODUCTION
A text-to-speech (TTS) system processes natural language text inputs to produce synthetic human-like speech outputs.
Typical TTS pipelines consist of a number of stages trained or designed independently – e.g. text normalisation, aligned linguistic featurisation, mel-spectrogram synthesis, and raw audio waveform synthesis (Taylor, 2009).
Although these pipelines have proven capable of realistic and high-ﬁdelity speech synthesis and enjoy wide real-world use today, these modular approaches come with a number of drawbacks.
They often require supervision at each stage, in some cases necessitating expensive “ground truth” annotations to guide the outputs of each stage, and sequential training of the stages.
Further, they are unable to reap the full potential rewards of data-driven “end-to-end" learning widely observed in a number of prediction and synthesis task domains across machine learning.
In this work, we aim to simplify the TTS pipeline and take on the challenging task of synthesising speech from text or phonemes in an end-to-end manner.
We propose EATS – End-to-end Adversarial
Text-to-Speech – generative models for TTS trained adversarially (Goodfellow et al., 2014) that operate on either pure text or raw (temporally unaligned) phoneme input sequences, and produce raw speech waveforms as output.
These models eliminate the typical intermediate bottlenecks present in most state-of-the-art TTS engines by maintaining learnt intermediate feature representations throughout the network.
Our speech synthesis models are composed of two high-level submodules, detailed in Section 2.
An aligner processes the raw input sequence and produces relatively low-frequency (200 Hz) aligned features in its own learnt, abstract feature space.
The features output by the aligner may be thought of as taking the place of the earlier stages of typical TTS pipelines – e.g., temporally aligned melspectrograms or linguistic features.
These features are then input to the decoder which upsamples the features from the aligner by 1D convolutions to produce 24 kHz audio waveforms.
By carefully designing the aligner and guiding training by a combination of adversarial feedback and domain-speciﬁc loss functions, we demonstrate that a TTS system can be learnt nearly end-to-end, Listen to our model reading this abstract at: https://deepmind.com/research/ publications/End-to-End-Adversarial-Text-to-Speech resulting in high-ﬁdelity natural-sounding speech approaching the state-of-the-art TTS systems.
Our main contributions include:
• A fully differentiable and efﬁcient feed-forward aligner architecture that predicts the duration of each input token and produces an audio-aligned representation.
• The use of ﬂexible dynamic time warping-based prediction losses to enforce alignment with input conditioning while allowing the model to capture the variability of timing in human speech.
• An overall system achieving a mean opinion score of 4 . 083 , approaching the state of the art from models trained using richer supervisory signals.
M ETHOD
Our goal is to learn a neural network (the generator) which maps an input sequence of characters or phonemes to raw audio at 24 kHz.
Beyond the vastly different lengths of the input and output signals, this task is also challenging because the input and output are not aligned, i.e. it is not known beforehand which output tokens each input token will correspond to.
To address these challenges, we divide the generator into two blocks: (i) the aligner, which maps the unaligned input sequence to a representation which is aligned with the output, but has a lower sample rate of 200 Hz; and (ii) the decoder, which upsamples the aligner’s output to the full audio frequency.
The entire generator architecture is differentiable, and is trained end to end.
Importantly, it is also a feed-forward convolutional network, which makes it well-suited for applications where fast batched inference is important: our EATS implementation generates speech at a speed of 200 × realtime on a single
NVIDIA V100 GPU (see Appendix A and Table 3 for details).
It is illustrated in Figure 1.
The generator is inspired by GAN-TTS (Bi ´ nkowski et al., 2020), a text-to-speech generative adversarial network operating on aligned linguistic features.
We employ the GAN-TTS generator as the decoder in our model, but instead of upsampling pre-computed linguistic features, its input comes from the aligner block.
We make it speaker-conditional by feeding in a speaker embedding s alongside the latent vector z , to enable training on a larger dataset with recordings from multiple speakers.
We also adopt the multiple random window discriminators (RWDs) from GAN-TTS, which have been proven effective for adversarial raw waveform modelling, and we preprocess real audio input by applying a simple µ -law transform.
Hence, the generator is trained to produce audio in the µ -law domain and we apply the inverse transformation to its outputs when sampling.
The loss function we use to train the generator is as follows:
[Equation]
where L G, adv is the adversarial loss, linear in the discriminators’ outputs, paired with the hinge loss (Lim & Ye, 2017; Tran et al., 2017) used as the discriminators’ objective, as used in GANTTS (Bi ´ nkowski et al., 2020).
The use of an adversarial (Goodfellow et al., 2014) loss is an advantage of our approach, as this setup allows for efﬁcient feed-forward training and inference, and such losses tend to be mode-seeking in practice, a useful behaviour in a strongly conditioned setting where realism is an important design goal, as in the case of text-to-speech.
In the remainder of this section, we describe the aligner network and the auxiliary prediction ( L (cid:48)(cid:48) pred ) and length ( L length ) losses in detail, and recap the components which were adopted from GAN-TTS.
2. A LIGNER
Given a token sequence x = ( x 1 , . . .
, x N ) of length N , we ﬁrst compute token representations h
= f ( x , z , s ) , where f is a stack of dilated convolutions (van den Oord et al., 2016) interspersed with batch normalisation (Ioffe & Szegedy, 2015) and ReLU activations.
The latents z and speaker embedding s modulate the scale and shift parameters of the batch normalisation layers (Dumoulin et al., 2017; De Vries et al., 2017).
We then predict the length for each input token individually: l n = g ( h n , z , s ) , where g is an MLP.
We use a ReLU nonlinearity at the output to ensure that the predicted lengths are non-negative.
We can then ﬁnd the predicted token end positions as a cumulative sum of the token lengths: e n = (cid:80) nm =1 l m , and the token centre positions as c n = e n − 12 l n .
Based on these predicted positions, we can interpolate the token representations into an audio-aligned representation at 200 Hz, a = ( a 1 , . .
. , a S ) , where S =
(cid:100)
e N (cid:101) is the total number of output time steps.
To compute a t , we obtain interpolation weights for the token representations h n using a softmax over the squared distance between t and c n , scaled by a temperature parameter σ 2 , which we set to 10 .(i.e.
a Gaussian kernel):
[Equation]
Using these weights, we can then compute a t =
(cid:80) Nn =1 w nt h n , which amounts to non-uniform interpolation.
By predicting token lengths and obtaining positions using cumulative summation, instead of predicting positions directly, we implicitly enforce monotonicity of the alignment.
Note that tokens which have a non-monotonic effect on prosody, such as punctuation, can still affect the entire utterance thanks to the stack of dilated convolutions f , whose receptive ﬁeld is large enough to allow for propagation of information across the entire token sequence.
The convolutions also ensure generalisation across different sequence lengths.
Appendix B includes pseudocode for the aligner.
2. W INDOWED GENERATOR TRAINING
Phonemizer
Dilated Conv Block ch =
256 x BN, ReLU, Conv ch = BN, ReLU, Conv ch =
t k s _ _ ɒ t _ ð n _ ə t .
æ
C a t s a t o n t
h
e m a t .
m c 2 c 3 cc N c
Næ æ c n = Σ m ≤ n l m - l n /logits grid x nt = - 𝜎 -2 ( c n - η - t )
𝜎 2 - temp. η - offset
Softmax along the horizontal ( n ) axis
Transpose
BN, ReLU
Speaker ID
Linear
Linear Linear
A l i g n e r l 2 l 3 ll N-1 l N l N-2 token lengths c n < η c n > η + T
Decoder
GAN-TTS
Generator η c N-1 token centres t
[Figure]
and is fed directly
Training examples vary widely in length, from about 1 to 20 seconds.
We cannot pad all sequences to a maximal length during training, as this would be wasteful and prohibitively expensive: 20 seconds of audio at 24 kHz correspond to 480,000 timesteps, which results in high memory requirements.
Instead, we randomly extract a 2 second window from each example, which we will refer to as a training window , by uniformly sampling a random offset η .
The aligner produces a 200 Hz audio-aligned representation for this window, which is then fed to the decoder (see Figure 1).
Note that we only need to compute a t for time steps t that fall within the sampled window, but we do have to compute the predicted token lengths l n for the entire input sequence.
During evaluation, we simply produce the audio-aligned representation for the full utterance and run the decoder on it, which is possible because it is fully convolutional.
2. A DVERSARIAL DISCRIMINATORS
Random window discriminators.
We use an ensemble of random window discriminators (RWDs) adopted from GAN-TTS.
Each RWD operates on audio fragments of different lengths, randomly sampled from the training window.
We use ﬁve RWDs with window sizes 240 , 480 , 960 , 1920 and 3600 .
This enables each RWD to operate at a different resolution.
Note thatsamples at 24 kHz corresponds to 150 ms of audio, so all RWDs operate on short timescales.
All RWDs in our model are unconditional with respect to text: they cannot access the text sequence or the aligner output.
(GAN-TTS uses 10 RWDs, including 5 conditioned on linguistic features which we omit.)
They are, however, conditioned on the speaker, via projection embedding (Miyato & Koyama, 2018).
Spectrogram discriminator.
We use an additional discriminator which operates on the full training window in the spectrogram domain.
We extract log-scaled mel-spectrograms from the audio signals and use the BigGAN-deep architecture (Brock et al., 2018), essentially treating the spectrograms as images.
The spectrogram discriminator also uses speaker identity through projection embedding.
Details on the spectrogram discriminator architecture are included in Appendix C.
2. S PECTROGRAM PREDICTION LOSS
In preliminary experiments, we discovered that adversarial feedback is insufﬁcient to learn alignment.
At the start of training, the aligner does not produce an accurate alignment, so the information in the input tokens is incorrectly temporally distributed.
This encourages the decoder to ignore the aligner output.
The unconditional discriminators provide no useful learning signal to correct this.
If we want to use conditional discriminators instead, we face a different problem: we do not have aligned ground truth.
Conditional discriminators also need an aligner module, which cannot function correctly at the start of training, effectively turning them into unconditional discriminators.
Although it should be possible in theory to train the discriminators’ aligner modules adversarially, we ﬁnd that this does not work in practice, and training gets stuck.
Instead, we propose to guide learning by using an explicit prediction loss in the spectrogram domain: we minimise the L 1 loss between the log-scaled mel-spectrograms of the generator output, and the corresponding ground truth training window.
This helps training to take off, and renders conditional discriminators unnecessary, simplifying the model.
Let S gen be the spectrogram of the generated audio, S gt the spectrogram of the corresponding ground truth, and S [ t, f ] the log-scaled magnitude at time step t and mel-frequency bin f .
Then the prediction loss is:
[Equation]
T and F are the total number of time steps and mel-frequency bins respectively.
Computing the prediction loss in the spectrogram domain, rather than the time domain, has the advantage of increased invariance to phase differences between the generated and ground truth signals, which are not perceptually salient.
Seeing as the spectrogram extraction operation has several hyperparameters and its implementation is not standardised, we provide the code we used for this in
Appendix D. We applied a small amount of jitter (by up to ± 60 samples at 24 kHz) to the ground truth waveform before computing S gt , which helped to reduce artifacts in the generated audio.
The inability to learn alignment from adversarial feedback alone is worth expanding on: likelihoodbased autoregressive models have no issues learning alignment, because they are able to beneﬁt from teacher forcing (Williams & Zipser, 1989) during training: the model is trained to perform next step prediction on each sequence step, given the preceding ground truth, and it is expected to infer alignment only one step at a time.
This is not compatible with feed-forward adversarial models however, so the prediction loss is necessary to bootstrap alignment learning for our model.
Note that although we make use of mel-spectrograms for training in L pred (and to compute the inputs for the spectrogram discriminator, Section 2.3), the generator itself does not produce spectrograms as part of the generation process.
Rather, its outputs are raw waveforms, and we convert these waveforms to spectrograms only for training (backpropagating gradients through the waveform to mel-spectrogram conversion operation).
2. D YNAMIC TIME WARPING
The spectrogram prediction loss incorrectly assumes that token lengths are deterministic.
We can relax the requirement that the generated and ground truth spectrograms are exactly aligned, by incorporating dynamic time warping (DTW)
(Sakoe, 1971; Sakoe & Chiba, 1978).
We calculate the prediction loss by iteratively ﬁnding a minimal-cost alignment path p between the generated and target spectrograms, S gen and S gt .
We start at the ﬁrst time step in both spectrograms: p gen , 1 = p gt , 1 = 1 .
At each iteration k , we take one of three possible actions:
go to the next time step in both S gen , S gt : p gen ,k +
=
p gen ,k + 1 , p gt ,k
+1 = p gt ,k + 1 ; 2. go to the next time step in S gt only: p gen ,k +1 = p gen ,k , p gt ,k +
=
p gt ,k + 1 ; 3. go to the next time step in S gen only: p gen ,k +
=
p gen ,k + 1 , p gt ,k
+1 = p gt ,k .
The resulting path is p = (cid:104) ( p gen , 1 , p gt , 1 ) , . . .
, ( p gen ,K p , p gt ,K p ) (cid:105) , where K p is the length.
Each action is assigned a cost based on the L 1 distance between S gen [ p gen ,k ] and S gt [ p gt ,k ] , and a warp penalty w which is incurred if we choose not to advance both spectrograms in lockstep (i.e. we are warping the spectrogram by taking action 2 or 3; we use w = 1 . 0 ).
The warp penalty thus encourages alignment paths that do not deviate too far from the identity alignment.
Let δ k be an indicator which is 1 for iterations where warping occurs, and 0 otherwise.
Then the total path cost c p is:
[Equation]
K p depends on the degree of warping ( T ≤ K p ≤ 2 T − 1 ).
The DTW prediction loss is then:
[Equation]
where P is the set of all valid paths.
p ∈ P only when p gen , 1 = p gt , 1 = 1 and p gen ,K p = p gt ,K p =
T , i.e. the ﬁrst and last timesteps of the spectrograms are aligned.
To ﬁnd the minimum, we use dynamic programming.
Figure 2 shows a diagram of an optimal alignment path between two sequences.
[Figure]
DTW is differentiable, but the minimum across all paths makes optimisation difﬁcult, because the gradient is propagated only through the minimal path.
We use a soft version of DTW instead (Cuturi &
Blondel, 2017), which replaces the minimum with the soft minimum:
[Equation]
where τ = 0 .
01 is a temperature parameter and the loss scale factor λ pred = 1 . 0 .
Note that the minimum operation is recovered by letting τ → 0 .
The resulting loss is a weighted aggregated cost across all paths, enabling gradient propagation through all feasible paths.
This creates a trade-off: a higher τ makes optimisation easier, but the resulting loss less accurately reﬂects the minimal path cost.
Pseudocode for the soft DTW procedure is provided in Appendix E.
By relaxing alignment in the prediction loss, the generator can produce waveforms that are not exactly aligned, without being heavily penalised for it.
This creates a synergy with the adversarial loss: instead of working against each other because of the rigidity of the prediction loss, the losses now cooperate to reward realistic audio generation with stochastic alignment.
Note that the prediction loss is computed on a training window, and not on full length utterances, so we still assume that the start and end points of the windows are exactly aligned.
While this might be incorrect, it does not seem to be much of a problem in practice.
2. A LIGNER LENGTH LOSS
To ensure that the model produces realistic token length predictions, we add a loss which encourages the predicted utterance length to be close to the ground truth length.
This length is found by summing all token length predictions.
Let L be the the number of time steps in the training utterance at 200 Hz, l n the predicted length of the n th token, and N the number of tokens, then the length loss is:
[Equation]
[Equation]
We use a scale factor λ length = 0 . 1 .
Note that we cannot match the predicted lengths l n to the ground truth lengths individually, because the latter are not available.
2. T EXT PRE - PROCESSING
Although our model works well with character input, we ﬁnd that sample quality improves signiﬁcantly using phoneme input instead.
This is not too surprising, given the heterogeneous way in which spellings map to phonemes, particularly in the English language.
Many character sequences also have special pronunciations, such as numbers, dates, units of measurement and website domains, and a very large training dataset would be required for the model to learn to pronounce these correctly.
Text normalisation (Zhang et al., 2019) can be applied beforehand to spell out these sequences as they are typically pronounced (e.g., 1976 could become nineteen seventy six ), potentially followed by conversion to phonemes.
We use an open source tool, phonemizer (Bernard, 2020), which performs partial normalisation and phonemisation (see Appendix F).
Finally, whether we train on text or phoneme input sequences, we pre- and post-pad the sequence with a special silence token (for training and inference), to allow the aligner to account for silence at the beginning and end of each utterance.
R ELATED WORK
Speech generation saw signiﬁcant quality improvements once treating it as a generative modelling problem became the norm (Zen et al., 2009; van den Oord et al., 2016).
Likelihood-based approaches dominate, but generative adversarial networks (GANs) (Goodfellow et al., 2014) have been making signiﬁcant inroads recently.
A common thread through most of the literature is a separation of the speech generation process into multiple stages: coarse-grained temporally aligned intermediate representations, such as mel-spectrograms, are used to divide the task into more manageable subproblems.
Many works focus exclusively on either spectrogram generation or vocoding (generating a waveform from a spectrogram).
Our work is different in this respect, and we will point out which stages of the generation process are addressed by each model.
In Appendix J, Table 6 we compare these methods in terms of the inputs and outputs to each stage of their pipelines.
Initially, most likelihood-based models for TTS were autoregressive (van den Oord et al., 2016; Mehri et al., 2017; Arik et al., 2017), which means that there is a sequential dependency between subsequent time steps of the produced output signal.
That makes these models impractical for real-time use, although this can be addressed with careful engineering (Kalchbrenner et al., 2018; Valin & Skoglund, 2019).
More recently, ﬂow-based models (Papamakarios et al., 2019) have been explored as a feed-forward alternative that enables fast inference (without sequential dependencies).
These can either be trained directly using maximum likelihood (Prenger et al., 2019;
Kim et al., 2019; Ping et al., 2019b), or through distillation from an autoregressive model (van den Oord et al., 2018; Ping et al., 2019a).
All of these models produce waveforms conditioned on an intermediate representation: either spectrograms or “linguistic features”, which contain temporally-aligned high-level information about the speech signal.
Spectrogram-conditioned waveform models are often referred to as vocoders .
A growing body of work has applied GAN (Goodfellow et al., 2014) variants to speech synthesis (Donahue et al., 2019).
An important advantage of adversarial losses for TTS is a focus on realism over diversity; the latter is less important in this setting.
This enables a more efﬁcient use of capacity compared to models trained with maximum likelihood.
MelGAN (Kumar et al., 2019) and
Parallel WaveGAN (Yamamoto et al., 2020) are adversarial vocoders, producing raw waveforms from mel-spectrograms.
Neekhara et al.
(2019) predict magnitude spectrograms from mel-spectrograms.
Most directly related to our work is GAN-TTS (Bi ´ nkowski et al., 2020), which produces waveforms conditioned on aligned linguistic features, and we build upon that work.
Another important line of work covers spectrogram generation from text.
Such models rely on a vocoder to convert the spectrograms into waveforms (for which one of the previously mentioned models could be used, or a traditional spectrogram inversion technique (Grifﬁn & Lim, 1984)).
Tacotron 1 & 2 (Wang et al., 2017; Shen et al., 2018), Deep Voice 2 & 3 (Gibiansky et al., 2017; Ping et al., 2018), TransformerTTS (Li et al., 2019), Flowtron (Valle et al., 2020), and VoiceLoop (Taigman et al., 2017) are autoregressive models that generate spectrograms or vocoder features frame by frame.
Guo et al.
(2019) suggest using an adversarial loss to reduce exposure bias (Bengio et al., 2015;
Ranzato et al., 2016) in such models.
MelNet (Vasquez & Lewis, 2019) is autoregressive over both time and frequency.
ParaNet
(Peng et al., 2019) and
FastSpeech (Ren et al., 2019) are nonautoregressive, but they require distillation (Hinton et al., 2015) from an autoregressive model.
Recent ﬂow-based approaches Flow-TTS (Miao et al., 2020) and Glow-TTS (Kim et al., 2020) are feedforward without requiring distillation.
Most spectrogram generation models require training of a custom vocoder model on generated spectrograms, because their predictions are imperfect and the vocoder needs to be able to compensate for this 2 .
Note that some of these works also propose new vocoder architectures in tandem with spectrogram generation models.
Unlike all of the aforementioned methods, as highlighted in Appendix J, Table 6, our model is a single feed-forward neural network, trained end-to-end in a single stage, which produces waveforms given character or phoneme sequences, and learns to align without additional supervision from auxiliary sources (e.g. temporally aligned linguistic features from an external model) or teacher forcing.
This simpliﬁes the training process considerably.
Char2wav (Sotelo et al., 2017) is ﬁnetuned end-to-end in the same fashion, but requires a pre-training stage with vocoder features used for intermediate supervision.
Spectrogram prediction losses have been used extensively for feed-forward audio prediction models (Yamamoto et al., 2019; 2020; Yang et al., 2020; Arık et al., 2018; Engel et al., 2020; Wang et al., 2019; Défossez et al., 2018).
We note that the L 1 loss we use (along with (Défossez et al., 2018)), is comparatively simple, as spectrogram losses in the literature tend to have separate terms penalising magnitudes, log-magnitudes and phase components, each with their own scaling factors, and often across multiple resolutions.
Dynamic time warping on spectrograms is a component of many speech recognition systems (Sakoe, 1971; Sakoe & Chiba, 1978), and has also been used for evaluation of
TTS systems (Sailor & Patil, 2014; Chevelu et al., 2015).
Cuturi & Blondel (2017) proposed the soft version of DTW we use in this work as a differentiable loss function for time series models.
Kim et al.
(2020) propose Monotonic Alignment Search (MAS), which relates to DTW in that both use dynamic programming to implicitly align sequences for TTS.
However, they have different goals:
MAS ﬁnds the optimal alignment between the text and a latent representation, whereas we use
DTW to relax the constraints imposed by our spectrogram prediction loss term.
Several mechanisms have been proposed to exploit monotonicity in tasks that require sequence alignment, including attention mechanisms (Graves, 2013; Zhang et al., 2018; Vasquez & Lewis, 2019;
He et al., 2019;
Raffel et al., 2017; Chiu & Raffel, 2018), loss functions (Graves et al., 2006; Graves, 2012) and search-based approaches (Kim et al., 2020).
For TTS, incorporating this constraint has been found to help generalisation to long sequences (Battenberg et al., 2020).
We incorporate monotonicity by using an interpolation mechanism, which is cheap to compute because it is not recurrent (unlike many monotonic attention mechanisms).
E VALUATION
In this section we discuss the setup and results of our empirical evaluation, describing the hyperparameter settings used for training and validating the architectural decisions and loss function components detailed in Section 2.
Our primary metric used to evaluate speech quality is the Mean Opinion Score (MOS) given by human raters, computed by taking the mean of 1-5 naturalness ratings given across 1000 held-out conditioning sequences.
In Appendix I we also report the Fréchet DeepSpeech Distance (FDSD), proposed by Bi ´ nkowski et al.
(2020) as a speech synthesis quality metric.
Appendix A reports training and evaluation hyperparameters we used for all experiments.
4. M ULTI - SPEAKER DATASET
We train all models on a private dataset that consists of high-quality recordings of human speech performed by professional voice actors, and corresponding text.
The voice pool consists of 69 female and male voices of North American English speakers, while the audio clips contain full sentences of lengths varying from less than 1 to 20 seconds at 24 kHz frequency.
Individual voices are unevenly distributed, accounting for from 15 minutes to over 51 hours of recorded speech, totalling 260.
hours.
At training time, we sample 2 second windows from the individual clips, post-padding those shorter than 2 seconds with silence.
For evaluation, we focus on the single most proliﬁc speaker in our dataset, with all our main MOS results reported with the model conditioned on that speaker ID, but also report MOS results for each of the top four speakers using our main multi-speaker model.
4. R ESULTS
In Table 1 we present quantitative results for our EATS model described in Section 2, as well as several ablations of the different model and learning signal components.
The architecture and training setup of each ablation is identical to our base EATS model except in terms of the differences described by the columns in Table 1.
Each ablation is “subtractive”, representing the full EATS system minus one particular feature.
Our main result achieved by the base multi-speaker model is a mean opinion score (MOS) of 4 . 083 .
Although it is difﬁcult to compare directly with prior results from the literature due to dataset differences, we nonetheless include MOS results from prior works (Bi´nkowski et al., 2020; van den Oord et al., 2016; 2018), with MOS in the 4.2 to 4.4+ range.
Compared to these prior models, which rely on aligned linguistic features, EATS uses substantially less supervision.
The No RWDs , No MelSpecD , and No Discriminators ablations all achieved substantially worse
MOS results than our proposed model, demonstrating the importance of adversarial feedback.
In particular, the No RWDs ablation, with an MOS of 2 . 526 , demonstrates the importance of the raw audio feedback, and removing RWDs signiﬁcantly degrades the high frequency components.
No
MelSpecD causes intermittent artifacts and distortion, and removing all discriminators results in audio that sounds robotic and distorted throughout.
The No L length and No L pred ablations result in a model that does not train at all.
Comparing our model with No DTW (MOS 3 . 559 ), the temporal
[Table]
[Table] All evaluations are done using our single multi-speaker EATS model. ﬂexibility provided by dynamic time warping signiﬁcantly improves ﬁdelity: removing it causes warbling and unnatural phoneme lengths.
No Phonemes is trained with raw character inputs and attains MOS 3 .
423 , due to occasional mispronunciations and unusual stress patterns.
No Mon.
Int.
uses an aligner with a transformer-based attention mechanism (described in Appendix G) in place of our monotonic interpolation architecture, which turns out to generalise poorly to long utterances (yielding MOS 3 . 551 ).
Finally, comparing against training with only a Single Speaker (MOS 3 . 829 ) shows that our EATS model beneﬁts from a much larger multi-speaker dataset, even though MOS is evaluated only on this same single speaker on which the ablation was solely trained.
Samples from each ablation are available at https://deepmind.com/research/publications/
End-to-End-Adversarial-Text-to-Speech .
We demonstrate that the aligner learns to use the latent vector z to vary the predicted token lengths in Appendix H. In Table 2 we present additional MOS results from our main multi-speaker EATS model for the four most proliﬁc speakers in our training data 3 .
MOS generally improves with more training data, although the correlation is imperfect (e.g., Speaker #3 achieves the highest MOS with only the third most training data).
D ISCUSSION
We have presented an adversarial approach to text-to-speech synthesis which can learn from a relatively weak supervisory signal – normalised text or phonemes paired with corresponding speech audio.
The speech generated by our proposed model matches the given conditioning texts and generalises to unobserved texts, with naturalness judged by human raters approaching state-of-the-art systems with multi-stage training pipelines or additional supervision.
The proposed system described in Section 2 is efﬁcient in both training and inference.
In particular, it does not rely on autoregressive sampling or teacher forcing, avoiding issues like exposure bias (Bengio et al., 2015; Ranzato et al., 2016) and reduced parallelism at inference time, or the complexities introduced by distillation to a more efﬁcient feed-forward model after the fact (van den Oord et al., 2018; Ping et al., 2019a).
While there remains a gap between the ﬁdelity of the speech produced by our method and the stateof-the-art systems, we nonetheless believe that the end-to-end problem setup is a promising avenue for future advancements and research in text-to-speech.
End-to-end learning enables the system as a whole to beneﬁt from large amounts of training data, freeing models to optimise their intermediate representations for the task at hand, rather than constraining them to work with the typical bottlenecks (e.g., mel-spectrograms, aligned linguistic features) imposed by most TTS pipelines today.
We see some evidence of this occurring in the comparison between our main result, trained using data from 69 speakers, against the Single Speaker ablation: the former is trained using roughly four times the data and synthesises more natural speech in the single voice on which the latter is trained.
Notably, our current approach does not attempt to address the text normalisation and phonemisation problems, relying on a separate, ﬁxed system for these aspects, while a fully end-to-end TTS system could operate on unnormalised raw text.
We believe that a fully data-driven approach could ultimately prevail even in this setup given sufﬁcient training data and model capacity. A CKNOWLEDGMENTS
The authors would like to thank Norman Casagrande, Yutian Chen, Aidan Clark, Kazuya Kawakami,
Pauline Luc, and many other colleagues at DeepMind for valuable discussions and input.
A H YPERPARAMETERS AND OTHER DETAILS
Our models are trained for 5 · 10 5 steps, where a single step consists of one discriminator update followed by one generator update, each using a minibatch size of 1024 , with batches sampled independently in each of these two updates.
Both updates are computed using the Adam optimizer (Kingma &
Ba, 2015) with β 1 = 0 and β 2 = 0 . 999 , and a learning rate of 10 − 3 with a cosine decay (Loshchilov & Hutter, 2017) schedule used such that the learning rate is 0 at step 500K. We apply spectral normalisation (Miyato et al., 2018) to the weights of the generator’s decoder module and to the discriminators (but not to the generator’s aligner module).
Parameters are initialised orthogonally and off-diagonal orthogonal regularisation with weight 10 − 4 is applied to the generator, following
BigGAN (Brock et al., 2018).
Minibatches are split over 64 or 128 cores (32 or 64 chips) of Google
Cloud TPU v3 Pods, which allows training of a single model within up to 58 hours.
We use crossreplica BatchNorm (Ioffe & Szegedy, 2015) to compute batch statistics aggregated across all devices.
Like in GAN-TTS (Bi´nkowski et al., 2020), our trained generator requires computation of standing statistics before sampling; i.e., accumulating batch norm statistics from 200 forward passes.
As in
GAN-TTS (Bi´nkowski et al., 2020) and BigGAN (Brock et al., 2018), we use an exponential moving average of the generator weights for inference, with a decay of 0 . 9999 .
Although GANs are known to exhibit stability issues sometimes, we found that EATS model training consistently converges.
Models were implemented using TensorFlow (Abadi et al., 2015) v1 framework and the Sonnet (Reynolds et al., 2017) neural network library.
We used the TF-Replicator (Buchlovsky et al., 2019) library for data parallel training over TPUs.
Inference speed.
In Table 3 we report benchmarks for EATS batched inference on two modern hardware platforms (Google Cloud TPU v3, NVIDIA V100 GPU, Intel Xeon E5-1650 CPU).
We ﬁnd that EATS can generate speech two orders of magnitude faster than realtime on a GPU or TPU, demonstrating the efﬁciency of our feed-forward model.
On the GPU, generating 2400 seconds ( minutes) of speech (80 utterances of 30 seconds each) takes 11.60 seconds on average (median), for a realtime factor of 206.8 × .
On TPU we observe a realtime factor of 157.2 × per chip (2 cores), or 78.6 × per core.
On
B A LIGNER PSEUDOCODE
In Figure 3 we present pseudocode for the EATS aligner described in Section 2.1.
C S PECTROGRAM DISCRIMINATOR ARCHITECTURE
In this Appendix we present details of the architecture of the spectrogram discriminator (Section 2.3).
The discriminator’s inputs are 47 × 80 × 1 images, produced by adding a channel dimension to the 47 × 80 output of the mel-spectrogram computation (Appendix D) from the length 48000 input waveforms (2 seconds of audio at 24 kHz).
Then, the architecture is like that of the BigGAN-deep (Brock et al., 2018) discriminator for 128 ×images (listed in BigGAN (Brock et al., 2018)
Appendix B, Table 7 (b)), but removing the ﬁrst two “ResBlocks” and the “Non-Local Block” (self-attention) – rows 2-4 in the architecture table (keeping row 1, the input convolution, and rows 5+ afterwards, as is).
This removes one 2 × 2 downsampling step as the resolution of the spectrogram inputs is smaller than the 128 × 128 images for which the
BigGAN-deep architecture was designed.
We set the channel width multiplier referenced in the table to ch =
[Table]
[Figure]
gen get mel spectrogram gen real spectrograms_for pred loss = get_mel_spectrogram(real waveforms jitter=
True)
D M EL - SPECTROGRAM COMPUTATION
In Figure 4 we include the TensorFlow (Abadi et al., 2015) code used to compute the mel-spectrograms fed into the spectrogram discriminator (Section 2.3) and the spectrogram prediction loss (Section 2.4).
Note that for use in the prediction losses L pred or L (cid:48)(cid:48) pred , we call this function with jitter=
True for real spectrograms and jitter=
False for generated spectrograms.
When used for the spectrogram discriminator inputs, we do not apply jitter to either real or generated spectrograms, setting jitter
=
False in both cases.
def soft_minimum(values, temperature): """Compute the soft minimum with the given temperature."
"" return -temperature * log(sum(exp(-values / temperature)))
def skew_matrix(x): """Skew a matrix so that the diagonals become the rows."
"" height, width = x.shape y
=
zeros(height + width - 1, width) for i in range(height + width - 1): for j in range(width): # Shift each column j down by j steps.
y[i, j] = x[clip(i - j, 0, height - 1), j] return y def spectrogram_dtw_error(spec_a, spec_b, warp_penalty=1.0, temperature=0.01): """Compute DTW error given a pair of spectrograms."
"" # Compute cost matrix.
diffs =
abs(spec_a[None, :, :] - spec_b[:, None, :]) costs = mean(diffs, axis=-1) # pairwise L1 cost, square the diffs for L2.
size = cost.shape[-1] # Initialise path costs.
[Table]
= INFINITY * ones(size + 1)
[Table]
# Aggregate path costs from cost[0, 0] to cost[ [Table] [Table]
[Table] = skew_ [Table] (cost)
# Shape is now (2 * size - 1, size).
for i in range(2 * size - 1): directions =
[path_cost_prev[:-1], path_cost[1:]+warp_penalty,path_cost[:-1]+warp_penalty] path_cost_next = cost[i] + soft_minimum(directions, temperature)
# Replace soft minimum with regular minimum for regular DTW.
path_cost_next = concatenate([[INFINITY], path_cost_next]) path_cost, path_cost_prev = path_cost_next, path_cost return path
[Figure]
E D YNAMIC TIME WARPING PSEUDOCODE
In Figure 5 we present pseudocode for the soft dynamic time warping (DTW) procedure we use in the spectrogram prediction loss L (cid:48)(cid:48) pred .
Note that the complexity of this implementation is quadratic.
It could be made more efﬁcient using
Itakura or Sakoe-Chiba bands (Itakura, 1975; Sakoe & Chiba, 1978), but we found that enabling or disabling DTW for the prediction loss did not meaningfully affect training time, so this optimisation is not necessary in practice.
Output symbol x ç ì j ; — ¡ r ~ "
Substitute symbol k k l j .
F T EXT PREPROCESSING
We use phonemizer (Bernard, 2020) (version 2.2) to perform partial normalisation and phonemisation of the input text (for all our results except for the No Phonemes ablation, where we use character sequences as input directly).
We used the espeak backend (with espeak-ng version 1.50), which produces phoneme sequences using the International Phonetic Alphabet (IPA).
We enabled the following options that phonemizer provides:
• with_stress , which includes primary and secondary stress marks in the output; • strip , which removes spurious whitespace; • preserve_punctuation , which ensures that punctuation is left unchanged.
This is important because punctuation can meaningfully affect prosody.
The phoneme sequences produced by phonemizer contain some rare symbols (usually in non-English words), which we replace with more frequent symbols.
The substitutions we perform are listed in
Table 4.
This results in a set of 51 distinct symbols.
The character sequence
Modern text-to-speech synthesis pipelines typically involve multiple processing stages.
becomes
[Equation]
dÄn
[Equation]
IpIkli
[Equation]
m­2ltIp [Equation] l pô"A:sEsIN st"eIdZ1z. G T RANSFORMER - BASED ATTENTION ALIGNER BASELINE
In this Appendix we describe our transformer-based attention aligner baseline, used in Sectionto compare against our monotonic interpolation-based aligner described in Section 2.1.
We use transformer attention (Vaswani et al., 2017) with output positional features as the queries, and a sum of input positional features and encoder output as the keys.
The encoder outputs are from the same dilated convolution stack as used in our EATS model, normalised using Layer Normalization (Ba et al., 2016) before input into the transformer.
We omit the fully-connected output layer following the attention mechanism.
Both sets of positional features use the sinusoidal encodings from Vaswani et al.
We use 4 heads with key and value dimensions of 64 per head.
Its outputs are taken as the audio-aligned feature representations, after which we apply Batch Normalisation and ReLU non-linearity before upsampling via the decoder.
[Figure]
[Figure]
H V ARIATION IN ALIGNMENT
To demonstrate that the aligner module makes use of the latent vector z to account for variations in token lengths
[Figure]
we generated 128 different renditions of the second sentence from the abstract: “In this work, we take on the challenging task of learning to synthesise speech from normalised text or phonemes in an end-to-end manner, resulting in models which operate directly on character or phoneme input sequences and produce raw speech audio outputs.” .
Figure 6 shows the positions of the tokens over time, with close-ups of the start and end of the sequence
[Figure]
to make the subtle variations in length more visible.
Figure 7 shows a histogram of the lengths of the generated utterances.
The variation is subtle (less than 2% for this utterance)
[Figure]
but noticeable.
Given that the training data consists of high-quality recordings of human speech performed by professional voice actors, only a modest degree of variation is to be expected.
[Table]
I E VALUATION WITH F RÉCHET D EEP S PEECH D ISTANCE
We found Fréchet DeepSpeech Distances (Bi ´ nkowski et al., 2020), both conditional and unconditional, unreliable in our setting.
Although they provided useful guidance at the early stages of model iteration – i.e., were able to clearly distinguish the models that do and do not train – FDSD scores of the models of reasonable quality were not in line with their Mean Opinion Scores, as shown for our ablations in
Table 5.
A possible reason for FDSD working less well in our setting is the fact that our models rely on features extracted from spectrograms similar to those computed at the DeepSpeech preprocessing stage.
As our models combine losses computed on raw audio and mel-spectrograms, it might be the case that the speech generated by some model is of lower quality, yet has convincing spectrograms.
Comparison of two of our ablations seems to afﬁrm this hypothesis: the No MelSpecD model achieves much higher MOS ( ≈ 3 . 5 ) than the No RWDs ablation ( ≈ 2 . 5 ) which is optimised only against spectrogram-based losses.
Their FDSDs, however, suggest the opposite ranking of these models.
Another potential cause for the discrepancy between MOS and FDSD is the difference in samples for which these scores were established.
While FDSD was computed on samples randomly held out from the training set, the MOS was computed on more challenging, often longer utterances.
As we did not have ground truth audio for the latter, we could not compute FDSD for these samples.
The sample sizes commonly used for the metrics based on Fréchet distance, e.g. (Heusel et al., 2017;
Kurach et al., 2019; Bi ´ nkowski et al., 2020), are also usually larger than the ones used for MOS testing (van den Oord et al., 2016; Bi´nkowski et al., 2020); we used 5,120 samples for FDSD and 1,000 for MOS.
We also note that conditional FDSD is not immediately applicable in our setting, as it requires ﬁxed length (two second) samples with aligned conditionings, while in our case there is no ﬁxed alignment between the ground truth characters and audio.
We hope that future research will revisit the challenge of automatic quantitative evaluation of text-tospeech models and produce a reliable quality metric for models operating in our current regime.
[Table]
J C OMPARISON OF TTS METHODS
In Table 6 we compare recent TTS approaches in terms of the inputs and outputs to each stage of the pipeline, and whether they are learnt in a single stage or multiple stages.
Differentiating EATS from each prior approach is the fact that it learns a feed-forward mapping from text/phonemes to audio end-to-end in a single stage, without requiring distillation or separate vocoder training.
The
ClariNet teacher model (Ping et al., 2019a) is also trained in a single stage, but it uses teacher forcing to achieve this, requiring the model to be autoregressive.
A separate distillation stage is necessary to obtain a feed-forward model in this case.