Tag Archives: Interspeech

Large-Scale Multilingual Speech Recognition with a Streaming End-to-End Model



Google's mission is not just to organize the world's information but to make it universally accessible, which means ensuring that our products work in as many of the world's languages as possible. When it comes to understanding human speech, which is a core capability of the Google Assistant, extending to more languages poses a challenge: high-quality automatic speech recognition (ASR) systems require large amounts of audio and text data — even more so as data-hungry neural models continue to revolutionize the field. Yet many languages have little data available.

We wondered how we could keep the quality of speech recognition high for speakers of data-scarce languages. A key insight from the research community was that much of the "knowledge" a neural network learns from audio data of a data-rich language is re-usable by data-scarce languages; we don't need to learn everything from scratch. This led us to study multilingual speech recognition, in which a single model learns to transcribe multiple languages.

In “Large-Scale Multilingual Speech Recognition with a Streaming End-to-End Model”, published at Interspeech 2019, we present an end-to-end (E2E) system trained as a single model, which allows for real-time multilingual speech recognition. Using nine Indian languages, we demonstrated a dramatic improvement in the ASR quality on several data-scarce languages, while still improving performance for the data-rich languages.

India: A Land of Languages
For this study, we focused on India, an inherently multilingual society where there are more than thirty languages with at least a million native speakers. Many of these languages overlap in acoustic and lexical content due to the geographic proximity of the native speakers and shared cultural history. Additionally, many Indians are bilingual or trilingual, making the use of multiple languages within a conversation a common phenomenon, and a natural case for training a single multilingual model. In this work, we combined nine primary Indian languages, namely Hindi, Marathi, Urdu, Bengali, Tamil, Telugu, Kannada, Malayalam and Gujarati.

A Low-latency All-neural Multilingual Model
Traditional ASR systems contain separate components for acoustic, pronunciation, and language models. While there have been attempts to make some or all of the traditional ASR components multilingual [1,2,3,4], this approach can be complex and difficult to scale. E2E ASR models combine all three components into a single neural network and promise scalability and ease of parameter sharing. Recent works have extended E2E models to be multilingual [1,2], but they did not address the need for real-time speech recognition, a key requirement for applications such as the Assistant, Voice Search and GBoard dictation. For this, we turned to recent research at Google that used a Recurrent Neural Network Transducer (RNN-T) model to achieve streaming E2E ASR. The RNN-T system outputs words one character at a time, just as if someone was typing in real time, however this was not multilingual. We built upon this architecture to develop a low-latency model for multilingual speech recognition.
[Left] A traditional monolingual speech recognizer comprising of Acoustic, Pronunciation and Language Models for each language. [Middle] A traditional multilingual speech recognizer where the Acoustic and Pronunciation model is multilingual, while the Language model is language-specific. [Right] An E2E multilingual speech recognizer where the Acoustic, Pronunciation and Language Model is combined into a single multilingual model.
Large-Scale Data Challenges
Using large-scale, real-world data for training a multilingual model is complicated by data imbalance. Given the steep skew in the distribution of speakers across the languages and speech product maturity, it is not surprising to have varying amounts of transcribed data available per language. As a result, a multilingual model can tend to be more influenced by languages that are over-represented in the training set. This bias is more prominent in an E2E model, which unlike a traditional ASR system, does not have access to additional in-language text data and learns lexical characteristics of the languages solely from the audio training data.
Histogram of training data for the nine languages showing the steep skew in the data available.
We addressed this issue with a few architectural modifications. First, we provided an extra language identifier input, which is an external signal derived from the language locale of the training data; i.e. the language preference set in an individual’s phone. This signal is combined with the audio input as a one-hot feature vector. We hypothesize that the model is able to use the language vector not only to disambiguate the language but also to learn separate features for separate languages, as needed, which helped with data imbalance.

Building on the idea of language-specific representations within the global model, we further augmented the network architecture by allocating extra parameters per language in the form of residual adapter modules. Adapters helped fine-tune a global model on each language while maintaining parameter efficiency of a single global model, and in turn, improved performance.
[Left] Multilingual RNN-T architecture with a language identifier. [Middle] Residual adapters inside the encoder. For a Tamil utterance, only the Tamil adapters are applied to each activation. [Right] Architecture details of the Residual Adapter modules. For more details please see our paper.
Putting all of these elements together, our multilingual model outperforms all the single-language recognizers, with especially large improvements in data-scarce languages like Kannada and Urdu. Moreover, since it is a streaming E2E model, it simplifies training and serving, and is also usable in low-latency applications like the Assistant. Building on this result, we hope to continue our research on multilingual ASRs for other language groups, to better assist our growing body of diverse users.

Acknowledgements
We would like to thank the following for their contribution to this research: Tara N. Sainath, Eugene Weinstein, Bo Li, Shubham Toshniwal, Ron Weiss, Bhuvana Ramabhadran, Yonghui Wu, Ankur Bapna, Zhifeng Chen, Seungji Lee, Meysam Bastani, Mikaela Grace, Pedro Moreno, Yanzhang (Ryan) He, Khe Chai Sim.

Source: Google AI Blog


Google at Interspeech 2019



This week, Graz, Austria hosts the 20th Annual Conference of the International Speech Communication Association (Interspeech 2019), one of the world‘s most extensive conferences on the research and engineering for spoken language processing. Over 2,000 experts in speech-related research fields gather to take part in oral presentations and poster sessions and to collaborate with streamed events across the globe.

As a Gold Sponsor of Interspeech 2019, we are excited to present 30 research publications, and demonstrate some of the impact speech technology has made in our products, from accessible, automatic video captioning to a more robust, reliable Google Assistant. If you’re attending Interspeech 2019, we hope that you’ll stop by the Google booth to meet our researchers and discuss projects and opportunities at Google that go into solving interesting problems for billions of people. Our researchers will also be on hand to discuss Google Cloud Text-to-Speech and Speech-to-text, demo Parrotron, and more. You can also learn more about the Google research being presented at Interspeech 2019 below (Google affiliations in blue).

Organizing Committee includes:
Michiel Bacchiani

Technical Program Committee includes:
Tara Sainath

Tutorials
Neural Machine Translation
Organizers include: Wolfgang Macherey, Yuan Cao

Accepted Publications
Building Large-Vocabulary ASR Systems for Languages Without Any Audio Training Data (link to appear soon)
Manasa Prasad, Daan van Esch, Sandy Ritchie, Jonas Fromseier Mortensen

Multi-Microphone Adaptive Noise Cancellation for Robust Hotword Detection (link to appear soon)
Yiteng Huang, Turaj Shabestary, Alexander Gruenstein, Li Wan

Direct Speech-to-Speech Translation with a Sequence-to-Sequence Model
Ye Jia, Ron Weiss, Fadi Biadsy, Wolfgang Macherey, Melvin Johnson, Zhifeng Chen, Yonghui Wu

Improving Keyword Spotting and Language Identification via Neural Architecture Search at Scale (link to appear soon)
Hanna Mazzawi, Javier Gonzalvo, Aleks Kracun, Prashant Sridhar, Niranjan Subrahmanya, Ignacio Lopez Moreno, Hyun Jin Park, Patrick Violette

Shallow-Fusion End-to-End Contextual Biasing (link to appear soon)
Ding Zhao, Tara Sainath, David Rybach, Pat Rondon, Deepti Bhatia, Bo Li, Ruoming Pang

VoiceFilter: Targeted Voice Separation by Speaker-Conditioned Spectrogram Masking
Quan Wang, Hannah Muckenhirn, Kevin Wilson, Prashant Sridhar, Zelin Wu, John Hershey, Rif Saurous, Ron Weiss, Ye Jia, Ignacio Lopez Moreno

SpecAugment: A Simple Data Augmentation Method for Automatic Speech Recognition
Daniel Park, William Chan, Yu Zhang, Chung-Cheng Chiu, Barret Zoph, Ekin Dogus Cubuk, Quoc Le

Two-Pass End-to-End Speech Recognition
Ruoming Pang, Tara Sainath, David Rybach, Yanzhang He, Rohit Prabhavalkar, Mirko Visontai, Qiao Liang, Trevor Strohman, Yonghui Wu, Ian McGraw, Chung-Cheng Chiu

On the Choice of Modeling Unit for Sequence-to-Sequence Speech Recognition
Kazuki Irie, Rohit Prabhavalkar, Anjuli Kannan, Antoine Bruguier, David Rybach, Patrick Nguyen

Contextual Recovery of Out-of-Lattice Named Entities in Automatic Speech Recognition (link to appear soon)
Jack Serrino, Leonid Velikovich, Petar Aleksic, Cyril Allauzen

Joint Speech Recognition and Speaker Diarization via Sequence Transduction
Laurent El Shafey, Hagen Soltau, Izhak Shafran

Personalizing ASR for Dysarthric and Accented Speech with Limited Data
Joel Shor, Dotan Emanuel, Oran Lang, Omry Tuval, Michael Brenner, Julie Cattiau, Fernando Vieira, Maeve McNally, Taylor Charbonneau, Melissa Nollstadt, Avinatan Hassidim, Yossi Matias

An Investigation Into On-Device Personalization of End-to-End Automatic Speech Recognition Models (link to appear soon)
Khe Chai Sim, Petr Zadrazil, Francoise Beaufays

Salient Speech Representations Based on Cloned Networks
Bastiaan Kleijn, Felicia Lim, Michael Chinen, Jan Skoglund

Cross-Lingual Consistency of Phonological Features: An Empirical Study (link to appear soon)
Cibu Johny, Alexander Gutkin, Martin Jansche

LibriTTS: A Corpus Derived from LibriSpeech for Text-to-Speech
Heiga Zen, Viet Dang, Robert Clark, Yu Zhang, Ron Weiss, Ye Jia, Zhifeng Chen, Yonghui Wu

Improving Performance of End-to-End ASR on Numeric Sequences
Cal Peyser, Hao Zhang, Tara Sainath, Zelin Wu

Developing Pronunciation Models in New Languages Faster by Exploiting Common Grapheme-to-Phoneme Correspondences Across Languages (link to appear soon)
Harry Bleyan, Sandy Ritchie, Jonas Fromseier Mortensen, Daan van Esch

Phoneme-Based Contextualization for Cross-Lingual Speech Recognition in End-to-End Models
Ke Hu, Antoine Bruguier, Tara Sainath, Rohit Prabhavalkar, Golan Pundak

Fréchet Audio Distance: A Reference-free Metric for Evaluating Music Enhancement Algorithms
Kevin Kilgour, Mauricio Zuluaga, Dominik Roblek, Matthew Sharifi

Learning to Speak Fluently in a Foreign Language: Multilingual Speech Synthesis and Cross-Language Voice Cloning
Yu Zhang, Ron Weiss, Heiga Zen, Yonghui Wu, Zhifeng Chen, RJ Skerry-Ryan, Ye Jia, Andrew Rosenberg, Bhuvana Ramabhadran

Sampling from Stochastic Finite Automata with Applications to CTC Decoding
Martin Jansche, Alexander Gutkin

Large-Scale Multilingual Speech Recognition with a Streaming End-to-End Model (link to appear soon)
Anjuli Kannan, Arindrima Datta, Tara Sainath, Eugene Weinstein, Bhuvana Ramabhadran, Yonghui Wu, Ankur Bapna, Zhifeng Chen, SeungJi Lee

A Real-Time Wideband Neural Vocoder at 1.6 kb/s Using LPCNet
Jean-Marc Valin, Jan Skoglund

Low-Dimensional Bottleneck Features for On-Device Continuous Speech Recognition
David Ramsay, Kevin Kilgour, Dominik Roblek, Matthew Sharif

Unified Verbalization for Speech Recognition & Synthesis Across Languages (link to appear soon)
Sandy Ritchie, Richard Sproat, Kyle Gorman, Daan van Esch, Christian Schallhart, Nikos Bampounis, Benoit Brard, Jonas Mortensen, Amelia Holt, Eoin Mahon

Better Morphology Prediction for Better Speech Systems (link to appear soon)
Dravyansh Sharma, Melissa Wilson, Antoine Bruguier

Dual Encoder Classifier Models as Constraints in Neural Text Normalization
Ajda Gokcen, Hao Zhang, Richard Sproat

Large-Scale Visual Speech Recognition
Brendan Shillingford, Yannis Assael, Matthew Hoffman, Thomas Paine, Cían Hughes, Utsav Prabhu, Hank Liao, Hasim Sak, Kanishka Rao, Lorrayne Bennett, Marie Mulville, Ben Coppin, Ben Laurie, Andrew Senior, Nando de Freitas

Parrotron: An End-to-End Speech-to-Speech Conversion Model and its Applications to Hearing-Impaired Speech and Speech Separation
Fadi Biadsy, Ron Weiss, Pedro Moreno, Dimitri Kanevsky, Ye Jia




Source: Google AI Blog


Project Euphonia’s Personalized Speech Recognition for Non-Standard Speech



The utility of technology is dependent on its accessibility. One key component of accessibility is automatic speech recognition (ASR), which can greatly improve the ability of those with speech impairments to interact with every-day smart devices. However, ASR systems are most often trained from 'typical' speech, which means that underrepresented groups, such as those with speech impairments or heavy accents, don't experience the same degree of utility. For example, amyotrophic lateral sclerosis (ALS) is a disease that can adversely affect a person’s speech—about 25% of people with ALS experiencing slurred speech as their first symptom. In addition, most people with ALS eventually lose the ability to walk, so being able to interact with automated devices from a distance can be very important. Yet current state-of-the-art ASR models can yield high word error rates (WER) for speakers with only a moderate speech impairment from ALS, effectively barring access to ASR reliant technologies.

In “Personalizing ASR for Dysarthric and Accented Speech with Limited Data,” to be presented at Interspeech 2019, we describe some of the research behind Project Euphonia, an ASR platform that performs speech-to-text transcription. This work presents an approach to improve ASR for people with ALS that may also be applicable to many other types of non-standard speech. Using a two-step training approach that starts with a baseline “standard” corpus and then fine-tunes the training with a personalized speech dataset, we have demonstrated significant improvements for speakers with atypical speech over current state-of-the-art models.

A Two-Phased Approach to Training
In order to create ASR models that work on non-standard speech, one needs to overcome two challenges. The first is that within a particular class of atypical speech, be it a regional accent or a speech impairment, for example, individuals can exhibit very different ways of speaking. Our approach deals with this sub-group heterogeneity by training the ASR model in two phases. We start with a high-quality ASR model trained on thousands of hours of standard speech and then we fine-tune parts of the model to an individual with non-standard speech. This approach is similar to that of Parrotron: both systems use end-to-end neural networks to help improve communication and accessibility, but Parrotron focuses exclusively on speech-to-speech, where a person’s speech is converted directly into synthesized speech, rather than text.

The second challenge arises from the difficulty in collecting enough data to train a state-of-the-art recognizer for individuals. Typical speech recognizers are trained on thousands of hours of speech from many different speakers. Acquiring this much data from a single speaker is nearly impossible, especially if the speaker may experience exhaustion from speaking due to a medical condition. Our approach overcomes this issue by first training a base model on a large corpus of typical speech, and then training a personalized model using a much smaller dataset with the targeted non-standard speech characteristics.

The Neural Network Architecture
When developing the models used for training data on atypical speech, we explored two different neural architectures. The first is the RNN-Transducer (RNN-T), a neural network architecture consisting of encoder and decoder networks that has shown good results on numerous ASR tasks. The encoder is bidirectional (i.e., it looks at the entire sentence at once in order to provide context), and thus it requires the entire audio sample to perform speech recognition.

The other architecture we explored was Listen, Attend, and Spell (LAS), which is an attention-based, sequence-to-sequence model that maps sequences of acoustic properties to sequences of languages. This model uses an encoder to convert the sequence of acoustic frames to a sequence of internal representations, and a decoder to convert the sequence of internal representations to linguistic output. The network produces “word pieces”, which are a linguistic representation between graphemes and words.
Comparison of the RNN-Transducer (left) and Listen, Attend, Spell (right) architectures. From Prabhavalkar et al. 2017.
We experimented with fine-tuning the state-of-the-art RNN-T and LAS base models on two types of non-standard speech. In partnership with the ALS Therapy Development Institute, we first collected about 36 hours of audio from 67 speakers who have ALS. The participants recorded themselves on their home computers using custom software while they read sentences from a very restricted language domain. Many phrases were single sentences with simple grammatical structure (e.g., “What time is the basketball game on tonight?”). This is in contrast with unrestricted language domains, which include domain-specific vocabulary (e.g., science talks) and complex language structure (e.g., a debate). The recordings did not include many of the filler words common in normal speech, such as “um” and “uh”.

We also tested accented speech, using the open source L2 Arctic dataset of non-native speech, which consists of 20 speakers with approximately 1 hour of speech per speaker. Each speaker recorded a set of 1150 utterances from the CMU Arctic prompts.

AudioEuphonia ModelStandard Speech Model
Did I have anything to say about it?Dictatorship angels to think about it
Come right back pleaseCameras object
Let’s try that againIt extracts
Turn it down a little bit pleaseTurning down a little bit please
The audio (left) are recordings of a speaker with ALS. The text transcriptions are output from the Euphonia model (center) and the Standard Speech model (right). Incorrectly transcribed text is underlined.
Results
The absolute word error rates on the language-restricted test set is shown below. There is an improvement over the baseline model for very non-standard speech (heavy accents and ALS speech below 3 on the ALS Functional Rating Scale) and moderate improvements in ALS speech that is similar to typical speech. The relative difference between the base model and the fine-tuned model demonstrates that the majority of the improvement comes from the fine-tuning process, except in the case of the RNN-T on the Arctic dataset, where the RNN-T baseline is already strong.
1 Non-native English speech from the L2-Arctic dataset.
2 Low FRS (ALS Functional Rating Scale) speech; intelligible with repeating (FRS 2); Speech combined with non-vocal communication (FRS 1).
3 FRS 3; detectable speech disturbance.
The RNN-T model achieved 91% of the improvement by fine-tuning just two layers, most of which are close to the input. On the accented dataset, fine-tuning the same two layers achieved 86% of the relative improvement compared to fine-tuning the entire network. This is consistent with previous speech work.

Most of the performance gains were achieved early in training. The models we trained were tested on a relatively limited domain of vocabulary and linguistic complexity, so the performance numbers are not necessarily related to how well the models perform on more general tasks. We hope that just fine-tuning part of the network allows it to retain the acoustic and linguistic information from the general speech model, while needing minimal modifications to adapt to a single new speaker. Future work will test this hypothesis.
Low FRS corresponds to the ALS speakers with low intelligibility (FRS 2, 1), while high FRS corresponds to ALS speakers with less severely impacted speech (FRS 3).
Understanding Model Behavior
To better understand how our models improved after fine-tuning, we looked at the pattern of phoneme mistakes. We started by comparing the distribution of phoneme mistakes made by the base ASR model on standard speech to the mistakes made on ALS speech. The SAMPA phonemes with the five largest differences between the ALS data and standard speech are p, U, f, k, and Z, which account for 20% of the deletion mistakes. Similarly, the n and m phonemes together account for 17% of the insertion / substitution mistakes. The same analysis on our fine-tuned models verifies that the unrecognized phoneme distribution is more similar to that of standard speech.

Our analysis shows that there are two aspects to every mistake: which phoneme the system doesn’t understand, and which phoneme the system thinks was said. Imagine having two systems with identical accuracy: one system always thinks that the f phoneme is actually the g phoneme, while another doesn't know what the f phoneme is and randomly guesses. These two systems will have identical performance and identical distributions of phoneme mistakes, but very different distributions of the predicted phoneme when a mistake is made. Surprisingly, ASR mistakes on ALS speech are far more similar to regular speech mistakes after Euphonia fine-tuning.
Deletion / substitution mistakes per SAMPA phoneme on ALS speech before fine-tuning, ALS speech after fine-tuning, and on typical speech (Librispeech dataset).
Future Work
In the future, we intend to explore additional techniques that can be helpful in the low data regime. We also hope to use phoneme mistakes to weight certain examples during training, or to pick training sentences for people with ALS to record that contain the most common phoneme mistakes. We would like to explore pooling data from multiple speakers with similar conditions.

We hope that continued research in this area will help voice interfaces become accessible to more people, especially those who need it most. One key component to this is collecting data. Anyone 18 or older can help us build better personalized models by donating audio data. If you’re interested, you can fill out this form to allow Google to contact you.

Acknowledgements
This work would not have been possible without the extraordinary effort and support of the ALS Therapy Development Institute and the ALS community, especially Fernando Vieira, Maeve McNally, Taylor Charbonneau, Melissa Nollstadt, and the individuals with ALS who kindly and patiently volunteered their audio. This work builds on the pioneering advances in speech recognition made by Google's speech team, in particular the recent development and deployment of end-to-end speech recognition models. We are grateful to the Google speech team for advice and collaboration, particularly to Anshuman Tripathi and Hasim Sak who guided us in training the initial models. We’d also like to thank Oran Lang, Omry Tuval, Michael Brenner, Julie Cattiau, Tara Sainath, Ding Zhao, Qiao Liang, Chung-Cheng Chiu, Dan Liebling, Ron Weiss, Anjuli Kannan, Dimitri Kanevsky, Ryan He, Gabor Simko, Benjamin Lee, Françoise Beaufays, Khe Chai Sim, Jimmy Tobin, Chet Gnegy, Jacqueline Huang, Ye Jia, Yu Zhang, Yonghui Wu, Michelle Ramanovich, Rus Heywood, Katrin Tomanek, Bob MacDonald, Pan-Pan Jiang, Ronnie Maor, Rif A. Saurous, Trevor Strohman, Dick Lyon, Avinatan Hassidim, Philip Nelson, and Yossi Matias for their technical contributions and project guidance.

Source: Google AI Blog