Tag Archives: Speech Recognition

Improving On-Device Speech Recognition with VoiceFilter-Lite

Voice assistive technologies, which enable users to employ voice commands to interact with their devices, rely on accurate speech recognition to ensure responsiveness to a specific user. But in many real-world use cases, the input to such technologies often consists of overlapping speech, which poses great challenges to many speech recognition algorithms. In 2018, we published a VoiceFilter system, which leverages Google’s Voice Match to personalize interaction with assistive technology by allowing people to enroll their voices.

While the VoiceFilter approach is highly successful, achieving a better source to distortion ratio (SDR) than conventional approaches, efficient on-device streaming speech recognition requires addressing restrictions such as model size, CPU and memory limitations, as well as battery usage considerations and latency minimization.

In “VoiceFilter-Lite: Streaming Targeted Voice Separation for On-Device Speech Recognition”, we present an update to VoiceFilter for on-device use that can significantly improve speech recognition in overlapping speech by leveraging the enrolled voice of a selected speaker. Importantly, this model can be easily integrated with existing on-device speech recognition applications, allowing the user to access voice assistive features under extremely noisy conditions even if an internet connection is unavailable. Our experiments show that a 2.2MB VoiceFilter-Lite model provides a 25.1% improvement to the word error rate (WER) on overlapping speech.

Improving On-Device Speech Recognition
While the original VoiceFilter system was very successful at separating a target speaker's speech signal from other overlapping sources, its model size, computational cost and latency are not feasible for speech recognition on mobile devices.

The new VoiceFilter-Lite system has been carefully designed to fit on-device applications. Instead of processing audio waveforms, VoiceFilter-Lite takes exactly the same input features as the speech recognition model (stacked log Mel-filterbanks), and directly enhances these features by filtering out components not belonging to the target speaker in real time. Together with several optimizations on network topologies, the number of runtime operations is drastically reduced. After quantizing the neural network with the TensorFlow Lite library, the model size is only 2.2 MB, which fits most on-device applications.

To train the VoiceFilter-Lite model, the filterbanks of the noisy speech are fed as input to the network together with an embedding vector that represents the identity of the target speaker (i.e., a d-vector). The network predicts a mask that is element-wise multiplied to the input to produce enhanced filterbanks. A loss function is defined to minimize the difference between the enhanced filterbanks and the filterbanks from the clean speech during training.

Model architecture of the VoiceFilter-Lite system.

VoiceFilter-Lite is a plug-and-play model, which allows the application in which it’s implemented to easily bypass it if the speaker did not enroll their voice. This also means that the speech recognition model and the VoiceFilter-Lite model can be separately trained and updated, which largely reduces engineering complexity in the deployment process.

As a plug-and-play model, VoiceFilter-Lite can be easily bypassed if the speaker did not enroll their voice.

Addressing the Challenge of Over-Suppression
When speech separation models are used for improving speech recognition, two types of error could occur: under-suppression, when the model fails to filter out noisy components from the signal; and over-suppression, when the model fails to preserve useful signal, resulting in some words being dropped from the recognized text. Over-suppression is especially problematic since modern speech recognition models are usually already trained with extensively augmented data (such as room simulation and SpecAugment), and thus are more robust to under-suppression.

VoiceFilter-Lite addresses the over-suppression issue with two novel approaches. First, it uses an asymmetric loss during the training process, such that the model is less tolerant to over-suppression than under-suppression. Second, it predicts the type of noise at runtime, and adaptively adjusts the suppression strength according to this prediction.

VoiceFilter-Lite adaptively applies stronger suppression strength when overlapping speech is detected.

With these two solutions, the VoiceFilter-Lite model retains great performance on streaming speech recognition for other scenarios, such as single-speaker speech under quiet or various noise conditions, while still providing significant improvement on overlapping speech. From our experiments, we observed a 25.1% improvement of word error rate after the 2.2MB VoiceFilter-Lite model is applied on additive overlapping speech. For reverberant overlapping speech, which is a more challenging task to simulate far-field devices such as smart home speakers, we also observed a 14.7% improvement of word error rate with VoiceFilter-Lite.

Future Work
While VoiceFilter-Lite has shown great promise for various on-device speech applications, we are also exploring several other directions to make VoiceFilter-Lite more useful. First, our current model is trained and evaluated with English speech only. We are excited about adopting the same technology to improve speech recognition for more languages. Second, we would like to directly optimize the speech recognition loss during the training of VoiceFilter-Lite, which can potentially further improve speech recognition beyond overlapping speech.

The research described in this post represents joint efforts from multiple teams within Google. Contributors include Quan Wang, Ignacio Lopez Moreno, Mert Saglam, Kevin Wilson, Alan Chiao, Renjie Liu, Yanzhang He, Wei Li, Jason Pelecanos, Philip Chao, Sinan Akay, John Han, Stephen Wu, Hannah Muckenhirn, Ye Jia, Zelin Wu, Yiteng Huang, Marily Nika, Jaclyn Konzelmann, Nino Tasca, and Alexander Gruenstein.

Source: Google AI Blog

Bringing Live Transcribe’s Speech Engine to Everyone

Earlier this year, Google launched Live Transcribe, an Android application that provides real-time automated captions for people who are deaf or hard of hearing. Through many months of user testing, we've learned that robustly delivering good captions for long-form conversations isn't so easy, and we want to make it easier for developers to build upon what we've learned. Live Transcribe's speech recognition is provided by Google's state-of-the-art Cloud Speech API, which under most conditions delivers pretty impressive transcript accuracy. However, relying on the cloud introduces several complications—most notably robustness to ever-changing network connections, data costs, and latency. Today, we are sharing our transcription engine with the world so that developers everywhere can build applications with robust transcription.

Those who have worked with our Cloud Speech API know that sending infinitely long streams of audio is currently unsupported. To help solve this challenge, we take measures to close and restart streaming requests prior to hitting the timeout, including restarting the session during long periods of silence and closing whenever there is a detected pause in the speech. Otherwise, this would result in a truncated sentence or word. In between sessions, we buffer audio locally and send it upon reconnection. This reduces the amount of text lost mid-conversation—either due to restarting speech requests or switching between wireless networks.

Endlessly streaming audio comes with its own challenges. In many countries, network data is quite expensive and in spots with poor internet, bandwidth may be limited. After much experimentation with audio codecs (in particular, we evaluated the FLAC, AMR-WB, and Opus codecs), we were able to achieve a 10x reduction in data usage without compromising accuracy. FLAC, a lossless codec, preserves accuracy completely, but doesn't save much data. It also has noticeable codec latency. AMR-WB, on the other hand, saves a lot of data, but delivers much worse accuracy in noisy environments. Opus was a clear winner, allowing data rates many times lower than most music streaming services while still preserving the important details of the audio signal—even in noisy environments. Beyond relying on codecs to keep data usage to a minimum, we also support using speech detection to close the network connection during extended periods of silence. That means if you accidentally leave your phone on and running Live Transcribe when nobody is around, it stops using your data.

Finally, we know that if you are relying on captions, you want them immediately, so we've worked hard to keep latency to a minimum. Though most of the credit for speed goes to the Cloud Speech API, Live Transcribe's final trick lies in our custom Opus encoder. At the cost of only a minor increase in bitrate, we see latency that is visually indistinguishable to sending uncompressed audio.

Today, we are excited to make all of this available to developers everywhere. We hope you'll join us in trying to build a world that is more accessible for everyone.

By Chet Gnegy, Alex Huang, and Ausmus Chang from the Live Transcribe Team

Parrotron: New Research into Improving Verbal Communication for People with Speech Impairments

Most people take for granted that when they speak, they will be heard and understood. But for the millions who live with speech impairments caused by physical or neurological conditions, trying to communicate with others can be difficult and lead to frustration. While there have been a great number of recent advances in automatic speech recognition (ASR; a.k.a. speech-to-text) technologies, these interfaces can be inaccessible for those with speech impairments. Further, applications that rely on speech recognition as input for text-to-speech synthesis (TTS) can exhibit word substitution, deletion, and insertion errors. Critically, in today’s technological environment, limited access to speech interfaces, such as digital assistants that depend on directly understanding one's speech, means being excluded from state-of-the-art tools and experiences, widening the gap between what those with and without speech impairments can access.

Project Euphonia has demonstrated that speech recognition models can be significantly improved to better transcribe a variety of atypical and dysarthric speech. Today, we are presenting Parrotron, an ongoing research project that continues and extends our effort to build speech technologies to help those with impaired or atypical speech to be understood by both people and devices. Parrotron consists of a single end-to-end deep neural network trained to convert speech from a speaker with atypical speech patterns directly into fluent synthesized speech, without an intermediate step of generating text—skipping speech recognition altogether. Parrotron’s approach is speech-centric, looking at the problem only from the point of view of speech signals—e.g., without visual cues such as lip movements. Through this work, we show that Parrotron can help people with a variety of atypical speech patterns—including those with ALS, deafness, and muscular dystrophy—to be better understood in both human-to-human interactions and by ASR engines.
The Parrotron Speech Conversion Model
Parrotron is an attention-based sequence-to-sequence model trained in two phases using parallel corpora of input/output speech pairs. First, we build a general speech-to-speech conversion model for standard fluent speech, followed by a personalization phase that adjusts the model parameters to the atypical speech patterns from the target speaker. The primary challenge in such a configuration lies in the collection of the parallel training data needed for supervised training, which consists of utterances spoken by many speakers and mapped to the same output speech content spoken by a single speaker. Since it is impractical to have a single speaker record the many hours of training data needed to build a high quality model, Parrotron uses parallel data automatically derived with a TTS system. This allows us to make use of a pre-existing anonymized, transcribed speech recognition corpus to obtain training targets.

The first training phase uses a corpus of ~30,000 hours that consists of millions of anonymized utterance pairs. Each pair includes a natural utterance paired with an automatically synthesized speech utterance that results from running our state-of-the-art Parallel WaveNet TTS system on the transcript of the first. This dataset includes utterances from thousands of speakers spanning hundreds of dialects/accents and acoustic conditions, allowing us to model a large variety of voices, linguistic and non-linguistic contents, accents, and noise conditions with “typical” speech all in the same language. The resulting conversion model projects away all non-linguistic information, including speaker characteristics, and retains only what is being said, not who, where, or how it is said. This base model is used to seed the second personalization phase of training.

The second training phase utilizes a corpus of utterance pairs generated in the same manner as the first dataset. In this case, however, the corpus is used to adapt the network to the acoustic/phonetic, phonotactic and language patterns specific to the input speaker, which might include, for example, learning how the target speaker alters, substitutes, and reduces or removes certain vowels or consonants. To model ALS speech characteristics in general, we use utterances taken from an ALS speech corpus derived from Project Euphonia. If instead we want to personalize the model for a particular speaker, then the utterances are contributed by that person. The larger this corpus is, the better the model is likely to be at correctly converting to fluent speech. Using this second smaller and personalized parallel corpus, we run the neural-training algorithm, updating the parameters of the pre-trained base model to generate the final personalized model.

We found that training the model with a multitask objective to predict the target phonemes while simultaneously generating spectrograms of the target speech led to significant quality improvements. Such a multitask trained encoder can be thought of as learning a latent representation of the input that maintains information about the underlying linguistic content.
Overview of the Parrotron model architecture. An input speech spectrogram is passed through encoder and decoder neural networks to generate an output spectrogram in a new voice.
Case Studies
To demonstrate a proof of concept, we worked with our fellow Google research scientist and mathematician Dimitri Kanevsky, who was born in Russia to Russian speaking, normal-hearing parents but has been profoundly deaf from a very young age. He learned to speak English as a teenager, by using Russian phonetic representations of English words, learning to pronounce English using transliteration into Russian (e.g., The quick brown fox jumps over the lazy dog => ЗИ КВИК БРАУН ДОГ ЖАМПС ОУВЕР ЛАЙЗИ ДОГ). As a result, Dimitri’s speech is substantially distinct from native English speakers, and can be challenging to comprehend for systems or listeners who are not accustomed to it.

Dimitri recorded a corpus of 15 hours of speech, which was used to adapt the base model to the nuances specific to his speech. The resulting Parrotron system helped him be better understood by both people and Google’s ASR system alike. Running Google’s ASR engine on the output of Parrotron significantly reduced the word error rate from 89% to 32%, on a held out test set from Dimitri. Below is an example of Parrotron’s successful conversion of input speech from Dimitri:

Input from Dimitri Audio
Output from Parrotron Audio

We also worked with Aubrie Lee, a Googler and advocate for disability inclusion, who has muscular dystrophy, a condition that causes progressive muscle weakness, and sometimes impacts speech production. Aubrie contributed 1.5 hours of speech, which has been instrumental in showing promising outcomes of the applicability of this speech-to-speech technology. Below is an example of Parrotron’s successful conversion of input speech from Aubrie:

Input from Aubrie Audio
Output from Parrotron Audio
Input from Aubrie Audio
Output from Parrotron Audio

We also tested Parrotron’s performance on speech from speakers with ALS by adapting the pretrained model on multiple speakers who share similar speech characteristics grouped together, rather than on a single speaker. We conducted a preliminary listening study and observed an increase in intelligibility when comparing natural ALS speech to the corresponding speech obtained from running the Parroton model, for the majority of our test speakers.

Cascaded Approach
Project Euphonia has built a personalized speech-to-text model that has reduced the word error rate for a deaf speaker from 89% to 25%, and ongoing research is also likely to improve upon these results. One could use such a speech-to-text model to achieve a similar goal as Parrotron by simply passing its output into a TTS system to synthesize speech from the result. In such a cascaded approach, however, the recognizer may choose an incorrect word (roughly 1 out 4 times, in this case)—i.e., it may yield words/sentences with unintended meaning and, as a result, the synthesized audio of these words would be far from the speaker’s intention. Given the end-to-end speech-to-speech training objective function of Parrotron, even when errors are made, the generated output speech is likely to sound acoustically similar to the input speech, and thus the speaker’s original intention is less likely to be significantly altered and it is often still possible to understand what is intended:

Input from Dimitri Audio
Output from Parrotron Audio
Input from Dimitri Audio
Output from Parrotron/Input to Assistant Audio
Output from Assistant Audio
Input from Aubrie Audio
Output from Parrotron Audio

Furthermore, since Parrotron is not strongly biased to producing words from a predefined vocabulary set, input to the model may contain completely new invented words, foreign words/names, and even nonsense words. We observe that feeding Arabic and Spanish utterances into the US-English Parrotron model often results in output which echoes the original speech content with an American accent, in the target voice. Such behavior is qualitatively different from what one would obtain by simply running an ASR followed by a TTS. Finally, by going from a combination of independently tuned neural networks to a single one, we also believe there are improvements and simplifications that could be substantial.

Parrotron makes it easier for users with atypical speech to talk to and be understood by other people and by speech interfaces, with its end-to-end speech conversion approach more likely to reproduce the user’s intended speech. More exciting applications of Parrotron are discussed in our paper and additional audio samples can be found on our github repository. If you would like to participate in this ongoing research, please fill out this short form and volunteer to record a set of phrases. We look forward to working with you!
This project was joint work between the Speech and Google Brain teams. Contributors include Fadi Biadsy, Ron Weiss, Pedro Moreno, Dimitri Kanevsky, Ye Jia, Suzan Schwartz, Landis Baker, Zelin Wu, Johan Schalkwyk, Yonghui Wu, Zhifeng Chen, Patrick Nguyen, Aubrie Lee, Andrew Rosenberg, Bhuvana Ramabhadran, Jason Pelecanos, Julie Cattiau, Michael Brenner, Dotan Emanuel and Joel Shor. Our data collection efforts have been vastly accelerated by our collaborations with ALS-TDI.

Source: Google AI Blog

An All-Neural On-Device Speech Recognizer

In 2012, speech recognition research showed significant accuracy improvements with deep learning, leading to early adoption in products such as Google's Voice Search. It was the beginning of a revolution in the field: each year, new architectures were developed that further increased quality, from deep neural networks (DNNs) to recurrent neural networks (RNNs), long short-term memory networks (LSTMs), convolutional networks (CNNs), and more. During this time, latency remained a prime focus — an automated assistant feels a lot more helpful when it responds quickly to requests.

Today, we're happy to announce the rollout of an end-to-end, all-neural, on-device speech recognizer to power speech input in Gboard. In our recent paper, "Streaming End-to-End Speech Recognition for Mobile Devices", we present a model trained using RNN transducer (RNN-T) technology that is compact enough to reside on a phone. This means no more network latency or spottiness — the new recognizer is always available, even when you are offline. The model works at the character level, so that as you speak, it outputs words character-by-character, just as if someone was typing out what you say in real-time, and exactly as you'd expect from a keyboard dictation system.
This video compares the production, server-side speech recognizer (left panel) to the new on-device recognizer (right panel) when recognizing the same spoken sentence. Video credit: Akshay Kannan and Elnaz Sarbar
A Bit of History
Traditionally, speech recognition systems consisted of several components - an acoustic model that maps segments of audio (typically 10 millisecond frames) to phonemes, a pronunciation model that connects phonemes together to form words, and a language model that expresses the likelihood of given phrases. In early systems, these components remained independently-optimized.

Around 2014, researchers began to focus on training a single neural network to directly map an input audio waveform to an output sentence. This sequence-to-sequence approach to learning a model by generating a sequence of words or graphemes given a sequence of audio features led to the development of "attention-based" and "listen-attend-spell" models. While these models showed great promise in terms of accuracy, they typically work by reviewing the entire input sequence, and do not allow streaming outputs as the input comes in, a necessary feature for real-time voice transcription.

Meanwhile, an independent technique called connectionist temporal classification (CTC) had helped halve the latency of the production recognizer at that time. This proved to be an important step in creating the RNN-T architecture adopted in this latest release, which can be seen as a generalization of CTC.

Recurrent Neural Network Transducers
RNN-Ts are a form of sequence-to-sequence models that do not employ attention mechanisms. Unlike most sequence-to-sequence models, which typically need to process the entire input sequence (the waveform in our case) to produce an output (the sentence), the RNN-T continuously processes input samples and streams output symbols, a property that is welcome for speech dictation. In our implementation, the output symbols are the characters of the alphabet. The RNN-T recognizer outputs characters one-by-one, as you speak, with white spaces where appropriate. It does this with a feedback loop that feeds symbols predicted by the model back into it to predict the next symbols, as described in the figure below.
Representation of an RNN-T, with the input audio samples, x, and the predicted symbols y. The predicted symbols (outputs of the Softmax layer) are fed back into the model through the Prediction network, as yu-1, ensuring that the predictions are conditioned both on the audio samples so far and on past outputs. The Prediction and Encoder Networks are LSTM RNNs, the Joint model is a feedforward network (paper). The Prediction Network comprises 2 layers of 2048 units, with a 640-dimensional projection layer. The Encoder Network comprises 8 such layers. Image credit: Chris Thornton
Training such models efficiently was already difficult, but with our development of a new training technique that further reduced the word error rate by 5%, it became even more computationally intensive. To deal with this, we developed a parallel implementation so the RNN-T loss function could run efficiently in large batches on Google's high-performance Cloud TPU v2 hardware. This yielded an approximate 3x speedup in training.

Offline Recognition
In a traditional speech recognition engine, the acoustic, pronunciation, and language models we described above are "composed" together into a large search graph whose edges are labeled with the speech units and their probabilities. When a speech waveform is presented to the recognizer, a "decoder" searches this graph for the path of highest likelihood, given the input signal, and reads out the word sequence that path takes. Typically, the decoder assumes a Finite State Transducer (FST) representation of the underlying models. Yet, despite sophisticated decoding techniques, the search graph remains quite large, almost 2GB for our production models. Since this is not something that could be hosted easily on a mobile phone, this method requires online connectivity to work properly.

To improve the usefulness of speech recognition, we sought to avoid the latency and inherent unreliability of communication networks by hosting the new models directly on device. As such, our end-to-end approach does not need a search over a large decoder graph. Instead, decoding consists of a beam search through a single neural network. The RNN-T we trained offers the same accuracy as the traditional server-based models but is only 450MB, essentially making a smarter use of parameters and packing information more densely. However, even on today's smartphones, 450MB is a lot, and propagating signals through such a large network can be slow.

We further reduced the model size by using the parameter quantization and hybrid kernel techniques we developed in 2016 and made publicly available through the model optimization toolkit in the TensorFlow Lite library. Model quantization delivered a 4x compression with respect to the trained floating point models and a 4x speedup at run-time, enabling our RNN-T to run faster than real time speech on a single core. After compression, the final model is 80MB.

Our new all-neural, on-device Gboard speech recognizer is initially being launched to all Pixel phones in American English only. Given the trends in the industry, with the convergence of specialized hardware and algorithmic improvements, we are hopeful that the techniques presented here can soon be adopted in more languages and across broader domains of application.

Raziel Alvarez, Michiel Bacchiani, Tom Bagby, Françoise Beaufays, Deepti Bhatia, Shuo-yiin Chang, Yanzhang He, Alex Gruenstein, Anjuli Kannan, Bo Li, Qiao Liang, Ian McGraw, Ruoming Pang, Rohit Prabhavalkar, Golan Pundak, Kanishka Rao, David Rybach, Tara Sainath, Haşim Sak, June Yuan Shangguan, Matt Shannon, Mohammadinamul Sheik, Khe Chai Sim, Gabor Simko, Trevor Strohman, Mirkó Visontai, Yonghui Wu, Ding Zhao, Dan Zivkovic.

Source: Google AI Blog

Teaching the Google Assistant to be Multilingual

Multilingual households are becoming increasingly common, with several sources [1][2][3] indicating that multilingual speakers already outnumber monolingual counterparts, and that this number will continue to grow. With this large and increasing population of multilingual users, it is more important than ever that Google develop products that can support multiple languages simultaneously to better serve our users.

Today, we’re launching multilingual support for the Google Assistant, which enables users to jump between two different languages across queries, without having to go back to their language settings. Once users select two of the supported languages, English, Spanish, French, German, Italian and Japanese, from there on out they can speak to the Assistant in either language and the Assistant will respond in kind. Previously, users had to choose a single language setting for the Assistant, changing their settings each time they wanted to use another language, but now, it’s a simple, hands-free experience for multilingual households.
The Google Assistant is now able to identify the language, interpret the query and provide a response using the right language without the user having to touch the Assistant settings.
Getting this to work, however, was not a simple feat. In fact, this was a multi-year effort that involved solving a lot of challenging problems. In the end, we broke the problem down into three discrete parts: Identifying Multiple Languages, Understanding Multiple Languages and Optimizing Multilingual Recognition for Google Assistant users.

Identifying Multiple Languages
People have the ability to recognize when someone is speaking another language, even if they do not speak the language themselves, just by paying attention to the acoustics of the speech (intonation, phonetic registry, etc). However, defining a computational framework for automatic spoken language recognition is challenging, even with the help of full automatic speech recognition systems1. In 2013, Google started working on spoken language identification (LangID) technology using deep neural networks [4][5]. Today, our state-of-the-art LangID models can distinguish between pairs of languages in over 2000 alternative language pairs using recurrent neural networks, a family of neural networks which are particularly successful for sequence modeling problems, such as those in speech recognition, voice detection, speaker recognition and others. One of the challenges we ran into was working with larger sets of audio — getting models that can automatically understanding multiple languages at scale, and hitting a quality standard that allowed those models to work properly.

Understanding Multiple Languages
To understand more than one language at once, multiple processes need to be run in parallel, each producing incremental results, allowing the Assistant not only to identify the language in which the query is spoken but also to parse the query to create an actionable command. For example, even for a monolingual environment, if a user asks to “set an alarm for 6pm”, the Google Assistant must understand that "set an alarm" implies opening the clock app, fulfilling the explicit parameter of “6pm” and additionally make the inference that the alarm should be set for today. To make this work for any given pair of supported languages is a challenge, as the Assistant executes the same work it does for the monolingual case, but now must additionally enable LangID, and not just one but two monolingual speech recognition systems simultaneously (we’ll explain more about the current two language limitation later in this post).

Importantly, the Google Assistant and other services that are referenced in the user’s query asynchronously generate real-time incremental results that need to be evaluated in a matter of milliseconds. This is accomplished with the help of an additional algorithm that ranks the transcription hypotheses provided by each of the two speech recognition systems using the probabilities of the candidate languages produced by LangID, our confidence on the transcription and the user’s preferences (such as favorite artists, for example).
Schematic of our multilingual speech recognition system used by the Google Assistant versus the standard monolingual speech recognition system. A ranking algorithm is used to select the best recognition hypotheses from two monolingual speech recognizer using relevant information about the user and the incremental langID results.
When the user stops speaking, the model has not only determined what language was being spoken, but also what was said. Of course, this process requires a sophisticated architecture that comes with an increased processing cost and the possibility of introducing unnecessary latency.

Optimizing Multilingual Recognition
To minimize these undesirable effects, the faster the system can make a decision about which language is being spoken, the better. If the system becomes certain of the language being spoken before the user finishes a query, then it will stop running the user’s speech through the losing recognizer and discard the losing hypothesis, thus lowering the processing cost and reducing any potential latency. With this in mind, we saw several ways of optimizing the system.

One use case we considered was that people normally use the same language throughout their query (which is also the language users generally want to hear back from the Assistant), with the exception of asking about entities with names in different languages. This means that, in most cases, focusing on the first part of the query allows the Assistant to make a preliminary guess of the language being spoken, even in sentences containing entities in a different language. With this early identification, the task is simplified by switching to a single monolingual speech recognizer, as we do for monolingual queries. Making a quick decision about how and when to commit to a single language, however, requires a final technological twist: specifically, we use a random forest technique that combines multiple contextual signals, such as the type of device being used, the number of speech hypotheses found, how often we receive similar hypotheses, the uncertainty of the individual speech recognizers, and how frequently each language is used.

An additional way we simplified and improved the quality of the system was to limit the list of candidate languages users can select. Users can choose two languages out of the six that our Home devices currently support, which will allow us to support the majority of our multilingual speakers. As we continue to improve our technology, however, we hope to tackle trilingual support next, knowing that this will further enhance the experience of our growing user base.

Bilingual to Trilingual
From the beginning, our goal has been to make the Assistant naturally conversational for all users. Multilingual support has been a highly-requested feature, and it’s something our team set its sights on years ago. But there aren’t just a lot of bilingual speakers around the globe today, we also want to make life a little easier for trilingual users, or families that live in homes where more than two languages are spoken.

With today’s update, we’re on the right track, and it was made possible by our advanced machine learning, our speech and language recognition technologies, and our team’s commitment to refine our LangID model. We’re now working to teach the Google Assistant how to process more than two languages simultaneously, and are working to add more supported languages in the future — stay tuned!

1 It is typically acknowledged that spoken language recognition is remarkably more challenging than text-based language identification where, relatively simple techniques based on dictionaries can do a good job. The time/frequency patterns of spoken words are difficult to compare, spoken words can be more difficult to delimit as they can be spoken without pause and at different paces and microphones may record background noise in addition to speech.

Source: Google AI Blog

Google Duplex: An AI System for Accomplishing Real World Tasks Over the Phone

A long-standing goal of human-computer interaction has been to enable people to have a natural conversation with computers, as they would with each other. In recent years, we have witnessed a revolution in the ability of computers to understand and to generate natural speech, especially with the application of deep neural networks (e.g., Google voice search, WaveNet). Still, even with today’s state of the art systems, it is often frustrating having to talk to stilted computerized voices that don't understand natural language. In particular, automated phone systems are still struggling to recognize simple words and commands. They don’t engage in a conversation flow and force the caller to adjust to the system instead of the system adjusting to the caller.

Today we announce Google Duplex, a new technology for conducting natural conversations to carry out “real world” tasks over the phone. The technology is directed towards completing specific tasks, such as scheduling certain types of appointments. For such tasks, the system makes the conversational experience as natural as possible, allowing people to speak normally, like they would to another person, without having to adapt to a machine.

One of the key research insights was to constrain Duplex to closed domains, which are narrow enough to explore extensively. Duplex can only carry out natural conversations after being deeply trained in such domains. It cannot carry out general conversations.

Here are examples of Duplex making phone calls (using different voices):
Duplex scheduling a hair salon appointment:
Duplex calling a restaurant:

While sounding natural, these and other examples are conversations between a fully automatic computer system and real businesses.

The Google Duplex technology is built to sound natural, to make the conversation experience comfortable. It’s important to us that users and businesses have a good experience with this service, and transparency is a key part of that. We want to be clear about the intent of the call so businesses understand the context. We’ll be experimenting with the right approach over the coming months.

Conducting Natural Conversations
There are several challenges in conducting natural conversations: natural language is hard to understand, natural behavior is tricky to model, latency expectations require fast processing, and generating natural sounding speech, with the appropriate intonations, is difficult.

When people talk to each other, they use more complex sentences than when talking to computers. They often correct themselves mid-sentence, are more verbose than necessary, or omit words and rely on context instead; they also express a wide range of intents, sometimes in the same sentence, e.g., “So umm Tuesday through Thursday we are open 11 to 2, and then reopen 4 to 9, and then Friday, Saturday, Sunday we... or Friday, Saturday we're open 11 to 9 and then Sunday we're open 1 to 9.”
Example of complex statement:

In natural spontaneous speech people talk faster and less clearly than they do when they speak to a machine, so speech recognition is harder and we see higher word error rates. The problem is aggravated during phone calls, which often have loud background noises and sound quality issues.

In longer conversations, the same sentence can have very different meanings depending on context. For example, when booking reservations “Ok for 4” can mean the time of the reservation or the number of people. Often the relevant context might be several sentences back, a problem that gets compounded by the increased word error rate in phone calls.

Deciding what to say is a function of both the task and the state of the conversation. In addition, there are some common practices in natural conversations — implicit protocols that include elaborations (“for next Friday” “for when?” “for Friday next week, the 18th.”), syncs (“can you hear me?”), interruptions (“the number is 212-” “sorry can you start over?”), and pauses (“can you hold? [pause] thank you!” different meaning for a pause of 1 second vs 2 minutes).

Enter Duplex
Google Duplex’s conversations sound natural thanks to advances in understanding, interacting, timing, and speaking.

At the core of Duplex is a recurrent neural network (RNN) designed to cope with these challenges, built using TensorFlow Extended (TFX). To obtain its high precision, we trained Duplex’s RNN on a corpus of anonymized phone conversation data. The network uses the output of Google’s automatic speech recognition (ASR) technology, as well as features from the audio, the history of the conversation, the parameters of the conversation (e.g. the desired service for an appointment, or the current time of day) and more. We trained our understanding model separately for each task, but leveraged the shared corpus across tasks. Finally, we used hyperparameter optimization from TFX to further improve the model.
Incoming sound is processed through an ASR system. This produces text that is analyzed with context data and other inputs to produce a response text that is read aloud through the TTS system.
Duplex handling interruptions:
Duplex elaborating:
Duplex responding to a sync:

Sounding Natural
We use a combination of a concatenative text to speech (TTS) engine and a synthesis TTS engine (using Tacotron and WaveNet) to control intonation depending on the circumstance.

The system also sounds more natural thanks to the incorporation of speech disfluencies (e.g. “hmm”s and “uh”s). These are added when combining widely differing sound units in the concatenative TTS or adding synthetic waits, which allows the system to signal in a natural way that it is still processing. (This is what people often do when they are gathering their thoughts.) In user studies, we found that conversations using these disfluencies sound more familiar and natural.

Also, it’s important for latency to match people’s expectations. For example, after people say something simple, e.g., “hello?”, they expect an instant response, and are more sensitive to latency. When we detect that low latency is required, we use faster, low-confidence models (e.g. speech recognition or endpointing). In extreme cases, we don’t even wait for our RNN, and instead use faster approximations (usually coupled with more hesitant responses, as a person would do if they didn’t fully understand their counterpart). This allows us to have less than 100ms of response latency in these situations. Interestingly, in some situations, we found it was actually helpful to introduce more latency to make the conversation feel more natural — for example, when replying to a really complex sentence.

System Operation
The Google Duplex system is capable of carrying out sophisticated conversations and it completes the majority of its tasks fully autonomously, without human involvement. The system has a self-monitoring capability, which allows it to recognize the tasks it cannot complete autonomously (e.g., scheduling an unusually complex appointment). In these cases, it signals to a human operator, who can complete the task.

To train the system in a new domain, we use real-time supervised training. This is comparable to the training practices of many disciplines, where an instructor supervises a student as they are doing their job, providing guidance as needed, and making sure that the task is performed at the instructor’s level of quality. In the Duplex system, experienced operators act as the instructors. By monitoring the system as it makes phone calls in a new domain, they can affect the behavior of the system in real time as needed. This continues until the system performs at the desired quality level, at which point the supervision stops and the system can make calls autonomously.

Benefits for Businesses and Users
Businesses that rely on appointment bookings supported by Duplex, and are not yet powered by online systems, can benefit from Duplex by allowing customers to book through the Google Assistant without having to change any day-to-day practices or train employees. Using Duplex could also reduce no-shows to appointments by reminding customers about their upcoming appointments in a way that allows easy cancellation or rescheduling.
Duplex calling a restaurant:

In another example, customers often call businesses to inquire about information that is not available online such as hours of operation during a holiday. Duplex can call the business to inquire about open hours and make the information available online with Google, reducing the number of such calls businesses receive, while at the same time, making the information more accessible to everyone. Businesses can operate as they always have, there’s no learning curve or changes to make to benefit from this technology.
Duplex asking for holiday hours:

For users, Google Duplex is making supported tasks easier. Instead of making a phone call, the user simply interacts with the Google Assistant, and the call happens completely in the background without any user involvement.
A user asks the Google Assistant for an appointment, which the Assistant then schedules by having Duplex call the business.
Another benefit for users is that Duplex enables delegated communication with service providers in an asynchronous way, e.g., requesting reservations during off-hours, or with limited connectivity. It can also help address accessibility and language barriers, e.g., allowing hearing-impaired users, or users who don’t speak the local language, to carry out tasks over the phone.

This summer, we’ll start testing the Duplex technology within the Google Assistant, to help users make restaurant reservations, schedule hair salon appointments, and get holiday hours over the phone.
Yaniv Leviathan, Google Duplex lead, and Matan Kalman, engineering manager on the project, enjoying a meal booked through a call from Duplex.
Duplex calling to book the above meal:

Allowing people to interact with technology as naturally as they interact with each other has been a long standing promise. Google Duplex takes a step in this direction, making interaction with technology via natural conversation a reality in specific scenarios. We hope that these technology advances will ultimately contribute to a meaningful improvement in people’s experience in day-to-day interactions with computers.

Source: Google AI Blog

Improving End-to-End Models For Speech Recognition

Traditional automatic speech recognition (ASR) systems, used for a variety of voice search applications at Google, are comprised of an acoustic model (AM), a pronunciation model (PM) and a language model (LM), all of which are independently trained, and often manually designed, on different datasets [1]. AMs take acoustic features and predict a set of subword units, typically context-dependent or context-independent phonemes. Next, a hand-designed lexicon (the PM) maps a sequence of phonemes produced by the acoustic model to words. Finally, the LM assigns probabilities to word sequences. Training independent components creates added complexities and is suboptimal compared to training all components jointly. Over the last several years, there has been a growing popularity in developing end-to-end systems, which attempt to learn these separate components jointly as a single system. While these end-to-end models have shown promising results in the literature [2, 3], it is not yet clear if such approaches can improve on current state-of-the-art conventional systems.

Today we are excited to share “State-of-the-art Speech Recognition With Sequence-to-Sequence Models [4],” which describes a new end-to-end model that surpasses the performance of a conventional production system [1]. We show that our end-to-end system achieves a word error rate (WER) of 5.6%, which corresponds to a 16% relative improvement over a strong conventional system which achieves a 6.7% WER. Additionally, the end-to-end model used to output the initial word hypothesis, before any hypothesis rescoring, is 18 times smaller than the conventional model, as it contains no separate LM and PM.

Our system builds on the Listen-Attend-Spell (LAS) end-to-end architecture, first presented in [2]. The LAS architecture consists of 3 components. The listener encoder component, which is similar to a standard AM, takes the a time-frequency representation of the input speech signal, x, and uses a set of neural network layers to map the input to a higher-level feature representation, henc. The output of the encoder is passed to an attender, which uses henc to learn an alignment between input features x and predicted subword units {yn, … y0}, where each subword is typically a grapheme or wordpiece. Finally, the output of the attention module is passed to the speller (i.e., decoder), similar to an LM, that produces a probability distribution over a set of hypothesized words.
Components of the LAS End-to-End Model.
All components of the LAS model are trained jointly as a single end-to-end neural network, instead of as separate modules like conventional systems, making it much simpler.
Additionally, because the LAS model is fully neural, there is no need for external, manually designed components such as finite state transducers, a lexicon, or text normalization modules. Finally, unlike conventional models, training end-to-end models does not require bootstrapping from decision trees or time alignments generated from a separate system, and can be trained given pairs of text transcripts and the corresponding acoustics.

In [4], we introduce a variety of novel structural improvements, including improving the attention vectors passed to the decoder and training with longer subword units (i.e., wordpieces). In addition, we also introduce numerous optimization improvements for training, including the use of minimum word error rate training [5]. These structural and optimization improvements are what accounts for obtaining the 16% relative improvement over the conventional model.

Another exciting potential application for this research is multi-dialect and multi-lingual systems, where the simplicity of optimizing a single neural network makes such a model very attractive. Here data for all dialects/languages can be combined to train one network, without the need for a separate AM, PM and LM for each dialect/language. We find that these models work well on 7 english dialects [6] and 9 Indian languages [7], while outperforming a model trained separately on each individual language/dialect.

While we are excited by our results, our work is not done. Currently, these models cannot process speech in real time [8, 9], which is a strong requirement for latency-sensitive applications such as voice search. In addition, these models still compare negatively to production when evaluated on live production data. Furthermore, our end-to-end model is learned on 22,000 audio-text pair utterances compared to a conventional system that is typically trained on significantly larger corpora. In addition, our proposed model is not able to learn proper spellings for rarely used words such as proper nouns, which is normally performed with a hand-designed PM. Our ongoing efforts are focused now on addressing these challenges.

This work was done as a strong collaborative effort between Google Brain and Speech teams. Contributors include Tara Sainath, Rohit Prabhavalkar, Bo Li, Kanishka Rao, Shankar Kumar, Shubham Toshniwal, Michiel Bacchiani and Johan Schalkwyk from the Speech team; as well as Yonghui Wu, Patrick Nguyen, Zhifeng Chen, Chung-cheng Chiu, Anjuli Kannan, Ron Weiss and Navdeep Jaitly from the Google Brain team. The work is described in more detail in papers [4-11]

[1] G. Pundak and T. N. Sainath, “Lower Frame Rate Neural Network Acoustic Models," in Proc. Interspeech, 2016.

[2] W. Chan, N. Jaitly, Q. V. Le, and O. Vinyals, “Listen, attend and spell,” CoRR, vol. abs/1508.01211, 2015

[3] R. Prabhavalkar, K. Rao, T. N. Sainath, B. Li, L. Johnson, and N. Jaitly, “A Comparison of Sequence-to-sequence Models for Speech Recognition,” in Proc. Interspeech, 2017.

[4] C.C. Chiu, T.N. Sainath, Y. Wu, R. Prabhavalkar, P. Nguyen, Z. Chen, A. Kannan, R.J. Weiss, K. Rao, K. Gonina, N. Jaitly, B. Li, J. Chorowski and M. Bacchiani, “State-of-the-art Speech Recognition With Sequence-to-Sequence Models,” submitted to ICASSP 2018.

[5] R. Prabhavalkar, T.N. Sainath, Y. Wu, P. Nguyen, Z. Chen, C.C. Chiu and A. Kannan, “Minimum Word Error Rate Training for Attention-based Sequence-to-Sequence Models,” submitted to ICASSP 2018.

[6] B. Li, T.N. Sainath, K. Sim, M. Bacchiani, E. Weinstein, P. Nguyen, Z. Chen, Y. Wu and K. Rao, “Multi-Dialect Speech Recognition With a Single Sequence-to-Sequence Model” submitted to ICASSP 2018.

[7] S. Toshniwal, T.N. Sainath, R.J. Weiss, B. Li, P. Moreno, E. Weinstein and K. Rao, “End-to-End Multilingual Speech Recognition using Encoder-Decoder Models”, submitted to ICASSP 2018.

[8] T.N. Sainath, C.C. Chiu, R. Prabhavalkar, A. Kannan, Y. Wu, P. Nguyen and Z. Chen, “Improving the Performance of Online Neural Transducer Models”, submitted to ICASSP 2018.

[9] D. Lawson*, C.C. Chiu*, G. Tucker*, C. Raffel, K. Swersky, N. Jaitly. “Learning Hard Alignments with Variational Inference”, submitted to ICASSP 2018.

[10] T.N. Sainath, R. Prabhavalkar, S. Kumar, S. Lee, A. Kannan, D. Rybach, V. Schogol, P. Nguyen, B. Li, Y. Wu, Z. Chen and C.C. Chiu, “No Need for a Lexicon? Evaluating the Value of the Pronunciation Lexica in End-to-End Models,” submitted to ICASSP 2018.

[11] A. Kannan, Y. Wu, P. Nguyen, T.N. Sainath, Z. Chen and R. Prabhavalkar. “An Analysis of Incorporating an External Language Model into a Sequence-to-Sequence Model,” submitted to ICASSP 2018.

Understanding Medical Conversations

Good documentation helps create good clinical care by communicating a doctor's thinking, their concerns, and their plans to the rest of the team. Unfortunately, physicians routinely spend more time doing documentation than doing what they love most — caring for patients. Part of the reason is that doctors spend ~6 hours in an 11-hour workday in the Electronic Health Records (EHR) on documentation.1 Consequently, one study found that more than half of surveyed doctors report at least one symptom of burnout.2

In order to help offload note-taking, many doctors have started using medical scribes as a part of their workflow. These scribes listen to the patient-doctor conversations and create notes for the EHR. According to a recent study, introducing scribes not only improved physician satisfaction, but also medical chart quality and accuracy.3 But the number of doctor-patient conversations that need a scribe is far beyond the capacity of people who are available for medical scribing.

We wondered: could the voice recognition technologies already available in Google Assistant, Google Home, and Google Translate be used to document patient-doctor conversations and help doctors and scribes summarize notes more quickly?
In “Speech Recognition for Medical Conversations”, we show that it is possible to build Automatic Speech Recognition (ASR) models for transcribing medical conversations. While most of the current ASR solutions in medical domain focus on transcribing doctor dictations (i.e., single speaker speech consisting of predictable medical terminology), our research shows that it is possible to build an ASR model which can handle multiple speaker conversations covering everything from weather to complex medical diagnosis.

Using this technology, we will start working with physicians and researchers at Stanford University, who have done extensive research on how scribes can improve physician satisfaction, to understand how deep learning techniques such as ASR can facilitate the scribing process of physician notes. In our pilot study, we investigate what types of clinically relevant information can be extracted from medical conversations to assist physicians in reducing their interactions with the EHR. The study is fully patient-consented and the content of the recording will be de-identified to protect patient privacy.

We hope these technologies will not only help return joy to practice by facilitating doctors and scribes with their everyday workload, but also help the patients get more dedicated and thorough medical attention, ideally, leading to better care.

1 http://www.annfammed.org/content/15/5/419.full
2 http://www.mayoclinicproceedings.org/article/S0025-6196%2815%2900716-8/abstract
3 http://www.annfammed.org/content/15/5/427.full

Launching the Speech Commands Dataset

At Google, we’re often asked how to get started using deep learning for speech and other audio recognition problems, like detecting keywords or commands. And while there are some great open source speech recognition systems like Kaldi that can use neural networks as a component, their sophistication makes them tough to use as a guide to a simpler tasks. Perhaps more importantly, there aren’t many free and openly available datasets ready to be used for a beginner’s tutorial (many require preprocessing before a neural network model can be built on them) or that are well suited for simple keyword detection.

To solve these problems, the TensorFlow and AIY teams have created the Speech Commands Dataset, and used it to add training* and inference sample code to TensorFlow. The dataset has 65,000 one-second long utterances of 30 short words, by thousands of different people, contributed by members of the public through the AIY website. It’s released under a Creative Commons BY 4.0 license, and will continue to grow in future releases as more contributions are received. The dataset is designed to let you build basic but useful voice interfaces for applications, with common words like “Yes”, “No”, digits, and directions included. The infrastructure we used to create the data has been open sourced too, and we hope to see it used by the wider community to create their own versions, especially to cover underserved languages and applications.

To try it out for yourself, download the prebuilt set of the TensorFlow Android demo applications and open up “TF Speech”. You’ll be asked for permission to access your microphone, and then see a list of ten words, each of which should light up as you say them.
The results will depend on whether your speech patterns are covered by the dataset, so it may not be perfect — commercial speech recognition systems are a lot more complex than this teaching example. But we’re hoping that as more accents and variations are added to the dataset, and as the community contributes improved models to TensorFlow, we’ll continue to see improvements and extensions.

You can also learn how to train your own version of this model through the new audio recognition tutorial on TensorFlow.org. With the latest development version of the framework and a modern desktop machine, you can download the dataset and train the model in just a few hours. You’ll also see a wide variety of options to customize the neural network for different problems, and to make different latency, size, and accuracy tradeoffs to run on different platforms.

We are excited to see what new applications people are able to build with the help of this dataset and tutorial, so I hope you get a chance to dive in and start recognizing!

* The architecture this network is based on is described in Convolutional Neural Networks for Small-footprint Keyword Spotting, presented at Interspeech 2015.

The Machine Intelligence Behind Gboard

Most people spend a significant amount of time each day using mobile-device keyboards: composing emails, texting, engaging in social media, and more. Yet, mobile keyboards are still cumbersome to handle. The average user is roughly 35% slower typing on a mobile device than on a physical keyboard. To change that, we recently provided many exciting improvements to Gboard for Android, working towards our vision of creating an intelligent mechanism that enables faster input while offering suggestions and correcting mistakes, in any language you choose.

With the realization that the way a mobile keyboard translates touch inputs into text is similar to how a speech recognition system translates voice inputs into text, we leveraged our experience in Speech Recognition to pursue our vision. First, we created robust spatial models that map fuzzy sequences of raw touch points to keys on the keyboard, just like acoustic models map sequences of sound bites to phonetic units. Second, we built a powerful core decoding engine based on finite state transducers (FST) to determine the likeliest word sequence given an input touch sequence. With its mathematical formalism and broad success in speech applications, we knew that an FST decoder would offer the flexibility needed to support a variety of complex keyboard input behaviors as well as language features. In this post, we will detail what went into the development of both of these systems.

Neural Spatial Models
Mobile keyboard input is subject to errors that are generally attributed to “fat finger typing” (or tracing spatially similar words in glide typing, as illustrated below) along with cognitive and motor errors (manifesting in misspellings, character insertions, deletions or swaps, etc). An intelligent keyboard needs to be able to account for these errors and predict the intended words rapidly and accurately. As such, we built a spatial model for Gboard that addresses these errors at the character level, mapping the touch points on the screen to actual keys.
Average glide trails for two spatially-similar words: “Vampire” and “Value”.
Up to recently, Gboard used a Gaussian model to quantify the probability of tapping neighboring keys and a rule-based model to represent cognitive and motor errors. These models were simple and intuitive, but they didn’t allow us to directly optimize metrics that correlate with better typing quality. Drawing on our experience with Voice Search acoustic models we replaced both the Gaussian and rule-based models with a single, highly efficient long short-term memory (LSTM) model trained with a connectionist temporal classification (CTC) criterion.

However, training this model turned out to be a lot more complicated than we had anticipated. While acoustic models are trained from human-transcribed audio data, one cannot easily transcribe millions of touch point sequences and glide traces. So the team exploited user-interaction signals, e.g. reverted auto-corrections and suggestion picks as negative and positive semi-supervised learning signals, to form rich training and test sets.
Raw data points corresponding to the word “could” (left), and normalized sampled trajectory with per-sample variances (right).
A plethora of techniques from the speech recognition literature was used to iterate on the NSM models to make them small and fast enough to be run on any device. The TensorFlow infrastructure was used to train hundreds of models, optimizing various signals surfaced by the keyboard: completions, suggestions, gliding, etc. After more than a year of work, the resulting models were about 6 times faster and 10 times smaller than the initial versions, they also showed about 15% reduction in bad autocorrects and 10% reduction in wrongly decoded gestures on offline datasets.

Finite-State Transducers
While the NSM uses spatial information to help determine what was tapped or swiped, there are additional constraints — lexical and grammatical — that can be brought to bear. A lexicon tells us what words occur in a language and a probabilistic grammar tells us what words are likely to follow other words. To encode this information we use finite-state transducers. FSTs have long been a key component of Google’s speech recognition and synthesis systems. They provide a principled way to represent various probabilistic models (lexicons, grammars, normalizers, etc) used in natural language processing together with the mathematical framework needed to manipulate, optimize, combine and search the models*.

In Gboard, a key-to-word transducer compactly represents the keyboard lexicon as shown in the figure below. It encodes the mapping from key sequences to words, allowing for alternative key sequences and optional spaces.
This transducer encodes “I”, “I’ve”, “If” along paths from the start state (the bold circle 1) to final states (the double circle states 0 and 1). Each arc is labeled with an input key (before the “:”) and a corresponding output word (after the “:”) where ε encodes the empty symbol. The apostrophe in “I’ve” can be omitted. The user may skip the space bar sometimes. To account for that, the space key transition between words in the transducer is optional. The ε and space back arcs allow accepting more than one word.
A probabilistic n-gram transducer is used to represent the language model for the keyboard. A state in the model represents an (up to) n-1 word context and an arc leaving that state is labeled with a successor word together with its probability of following that context (estimated from textual data). These, together with the spatial model that gives the likelihoods of sequences of key touches (discrete tap entries or continuous gestures in glide typing), are combined and explored with a beam search.

Generic FST principles, such as streaming, support for dynamic models, etc took us a long way towards building a new keyboard decoder, but several new functionalities also had to be added. When you speak, you don’t need the decoder to complete your words or guess what you will say next to save you a few syllables; but when you type, you appreciate the help of word completions and predictions. Also, we wanted the keyboard to provide seamless multilingual support, as shown below.
Trilingual input typing in Gboard.
It was a complex effort to get our new decoder off the ground, but the principled nature of FSTs has many benefits. For example, supporting transliterations for languages like Hindi is just a simple extension of the generic decoder.

Transliteration Models
In many languages with complex scripts, romanization systems have been developed to map characters into the Latin alphabet, often according to their phonetic pronunciations. For example, the Pinyin “xièxiè” corresponds to the Chinese characters “谢谢” (“thank you”). A Pinyin keyboard allows users to conveniently type words on a QWERTY layout and have them automatically “translated” into the target script. Likewise, a transliterated Hindi keyboard allows users to type “daanth” for “दांत” (teeth). Whereas Pinyin is an agreed-upon romanization system, Hindi transliterations are more fuzzy; for example “daant” would be a valid alternative for “दांत”.
Transliterated glide input for Hindi.
Just as we have a transducer mapping from letter sequences to words (a lexicon) and a weighted language model automaton providing probabilities for word sequences, we built weighted transducer mappings between Latin key sequences and target script symbol sequences for 22 Indic languages. Some languages have multiple writing systems (Bodo for example can be written in the Bengali or Devanagari scripts) so between transliterated and native layouts, we built 57 new input methods in just a few months.

The general nature of the FST decoder let us leverage all the work we had done to support completions, predictions, glide typing and many UI features with no extra effort, allowing us to offer a rich experience to our Indian users right from the start.

A More Intelligent Keyboard
All in all, our recent work cut the decoding latency by 50%, reduced the fraction of words users have to manually correct by more than 10%, allowed us to launch transliteration support for the 22 official languages of India, and enabled many new features you may have noticed.

While we hope that these recent changes improve your typing experience, we recognize that on-device typing is by no means solved. Gboard can still make suggestions that seem nonintuitive or of low utility and gestures can still be decoded to words a human would never pick. However, our shift towards powerful machine intelligence algorithms has opened new spaces that we’re actively exploring to make more useful tools and products for our users worldwide.

This work was done by Cyril Allauzen, Ouais Alsharif, Lars Hellsten, Tom Ouyang, Brian Roark and David Rybach, with help from Speech Data Operation team. Special thanks go to Johan Schalkwyk and Corinna Cortes for their support.

* The toolbox of relevant algorithms is available in the OpenFst open-source library.