![](https://csdnimg.cn/release/download_crawler_static/33495660/bg3.jpg)
speaker
reference
waveform
Speaker
Encoder
grapheme or
phoneme
sequence
Encoder
concat
Attention Decoder
Synthesizer
Vocoder
waveform
speaker
embedding
log-mel
spectrogram
Figure 1: Model overview. Each of the three components are trained independently.
signal, (2) a sequence-to-sequence synthesizer, based on [
15
], which predicts a mel spectrogram from
a sequence of grapheme or phoneme inputs, conditioned on the speaker embedding vector, and (3) an
autoregressive WaveNet [
19
] vocoder, which converts the spectrogram into time domain waveforms.
1
2.1 Speaker encoder
The speaker encoder is used to condition the synthesis network on a reference speech signal from the
desired target speaker. Critical to good generalization is the use of a representation which captures the
characteristics of different speakers, and the ability to identify these characteristics using only a short
adaptation signal, independent of its phonetic content and background noise. These requirements are
satisfied using a speaker-discriminative model trained on a text-independent speaker verification task.
We follow [
22
], which proposed a highly scalable and accurate neural network framework for speaker
verification. The network maps a sequence of log-mel spectrogram frames computed from a speech
utterance of arbitrary length, to a fixed-dimensional embedding vector, known as d-vector [
20
,
9
]. The
network is trained to optimize a generalized end-to-end speaker verification loss, so that embeddings
of utterances from the same speaker have high cosine similarity, while those of utterances from
different speakers are far apart in the embedding space. The training dataset consists of speech audio
examples segmented into 1.6 seconds and associated speaker identity labels; no transcripts are used.
Input 40-channel log-mel spectrograms are passed to a network consisting of a stack of 3 LSTM
layers of 768 cells, each followed by a projection to 256 dimensions. The final embedding is created
by
L
2
-normalizing the output of the top layer at the final frame. During inference, an arbitrary length
utterance is broken into 800ms windows, overlapped by 50%. The network is run independently on
each window, and the outputs are averaged and normalized to create the final utterance embedding.
Although the network is not optimized directly to learn a representation which captures speaker
characteristics relevant to synthesis, we find that training on a speaker discrimination task leads to an
embedding which is directly suitable for conditioning the synthesis network on speaker identity.
2.2 Synthesizer
We extend the recurrent sequence-to-sequence with attention Tacotron 2 architecture [
15
] to support
multiple speakers following a scheme similar to [
8
]. An embedding vector for the target speaker is
concatenated with the synthesizer encoder output at each time step. In contrast to [
8
], we find that
simply passing embeddings to the attention layer, as in Figure 1, converges across different speakers.
We compare two variants of this model, one which computes the embedding using the speaker
encoder, and a baseline which optimizes a fixed embedding for each speaker in the training set,
essentially learning a lookup table of speaker embeddings similar to [8, 13].
The synthesizer is trained on pairs of text transcript and target audio. At the input, we map the text to
a sequence of phonemes, which leads to faster convergence and improved pronunciation of rare words
and proper nouns. The network is trained in a transfer learning configuration, using a pretrained
speaker encoder (whose parameters are frozen) to extract a speaker embedding from the target audio,
i.e. the speaker reference signal is the same as the target speech during training. No explicit speaker
identifier labels are used during training.
Target spectrogram features are computed from 50ms windows computed with a 12.5ms step, passed
through an 80-channel mel-scale filterbank followed by log dynamic range compression. We extend
[
15
] by augmenting the
L
2
loss on the predicted spectrogram with an additional
L
1
loss. In practice,
1
See https://google.github.io/tacotron/publications/speaker_adaptation for samples.
3