Tag Archives: Automatic Speech Recognition

Responsible AI at Google Research: AI for Social Good

Posted by Jimmy Tobin and Katrin Tomanek, Software Engineers, Google Research, AI for Social Good

Google’s AI for Social Good team consists of researchers, engineers, volunteers, and others with a shared focus on positive social impact. Our mission is to demonstrate AI’s societal benefit by enabling real-world value, with projects spanning work in public health, accessibility, crisis response, climate and energy, and nature and society. We believe that the best way to drive positive change in underserved communities is by partnering with change-makers and the organizations they serve.

In this blog post we discuss work done by Project Euphonia, a team within AI for Social Good, that aims to improve automatic speech recognition (ASR) for people with disordered speech. For people with typical speech, an ASR model’s word error rate (WER) can be less than 10%. But for people with disordered speech patterns, such as stuttering, dysarthria and apraxia, the WER could reach 50% or even 90% depending on the etiology and severity. To help address this problem, we worked with more than 1,000 participants to collect over 1,000 hours of disordered speech samples and used the data to show that ASR personalization is a viable avenue for bridging the performance gap for users with disordered speech. We've shown that personalization can be successful with as little as 3-4 minutes of training speech using layer freezing techniques.

This work led to the development of Project Relate for anyone with atypical speech who could benefit from a personalized speech model. Built in partnership with Google’s Speech team, Project Relate enables people who find it hard to be understood by other people and technology to train their own models. People can use these personalized models to communicate more effectively and gain more independence. To make ASR more accessible and usable, we describe how we fine-tuned Google’s Universal Speech Model (USM) to better understand disordered speech out of the box, without personalization, for use with digital assistant technologies, dictation apps, and in conversations.


Addressing the challenges

Working closely with Project Relate users, it became clear that personalized models can be very useful, but for many users, recording dozens or hundreds of examples can be challenging. In addition, the personalized models did not always perform well in freeform conversation.

To address these challenges, Euphonia’s research efforts have been focusing on speaker independent ASR (SI-ASR) to make models work better out of the box for people with disordered speech so that no additional training is necessary.


Prompted Speech dataset for SI-ASR

The first step in building a robust SI-ASR model was to create representative dataset splits. We created the Prompted Speech dataset by splitting the Euphonia corpus into train, validation and test portions, while ensuring that each split spanned a range of speech impairment severity and underlying etiology and that no speakers or phrases appeared in multiple splits. The training portion consists of over 950k speech utterances from over 1,000 speakers with disordered speech. The test set contains around 5,700 utterances from over 350 speakers. Speech-language pathologists manually reviewed all of the utterances in the test set for transcription accuracy and audio quality.


Real Conversation test set

Unprompted or conversational speech differs from prompted speech in several ways. In conversation, people speak faster and enunciate less. They repeat words, repair misspoken words, and use a more expansive vocabulary that is specific and personal to themselves and their community. To improve a model for this use case, we created the Real Conversation test set to benchmark performance.

The Real Conversation test set was created with the help of trusted testers who recorded themselves speaking during conversations. The audio was reviewed, any personally identifiable information (PII) was removed, and then that data was transcribed by speech-language pathologists. The Real Conversation test set contains over 1,500 utterances from 29 speakers.


Adapting USM to disordered speech

We then tuned USM on the training split of the Euphonia Prompted Speech set to improve its performance on disordered speech. Instead of fine-tuning the full model, our tuning was based on residual adapters, a parameter-efficient tuning approach that adds tunable bottleneck layers as residuals between the transformer layers. Only these layers are tuned, while the rest of the model weights are untouched. We have previously shown that this approach works very well to adapt ASR models to disordered speech. Residual adapters were only added to the encoder layers, and the bottleneck dimension was set to 64.


Results

To evaluate the adapted USM, we compared it to older ASR models using the two test sets described above. For each test, we compare adapted USM to the pre-USM model best suited to that task: (1) For short prompted speech, we compare to Google’s production ASR model optimized for short form ASR; (2) for longer Real Conversation speech, we compare to a model trained for long form ASR. USM improvements over pre-USM models can be explained by USM’s relative size increase, 120M to 2B parameters, and other improvements discussed in the USM blog post.

Model word error rates (WER) for each test set (lower is better).

We see that the USM adapted with disordered speech significantly outperforms the other models. The adapted USM’s WER on Real Conversation is 37% better than the pre-USM model, and on the Prompted Speech test set, the adapted USM performs 53% better.

These findings suggest that the adapted USM is significantly more usable for an end user with disordered speech. We can demonstrate this improvement by looking at transcripts of Real Conversation test set recordings from a trusted tester of Euphonia and Project Relate (see below).


Audio1    Ground Truth    Pre-USM ASR    Adapted USM
                    
   I now have an Xbox adaptive controller on my lap.    i now have a lot and that consultant on my mouth    i now had an xbox adapter controller on my lamp.
                    
   I've been talking for quite a while now. Let's see.    quite a while now    i've been talking for quite a while now.
Example audio and transcriptions of a trusted tester’s speech from the Real Conversation test set.

A comparison of the Pre-USM and adapted USM transcripts revealed some key advantages:

  • The first example shows that Adapted USM is better at recognizing disordered speech patterns. The baseline misses key words like “XBox” and “controller” that are important for a listener to understand what they are trying to say.
  • The second example is a good example of how deletions are a primary issue with ASR models that are not trained with disordered speech. Though the baseline model did transcribe a portion correctly, a large part of the utterance was not transcribed, losing the speaker’s intended message.

Conclusion

We believe that this work is an important step towards making speech recognition more accessible to people with disordered speech. We are continuing to work on improving the performance of our models. With the rapid advancements in ASR, we aim to ensure people with disordered speech benefit as well.


Acknowledgements

Key contributors to this project include Fadi Biadsy, Michael Brenner, Julie Cattiau, Richard Cave, Amy Chung-Yu Chou, Dotan Emanuel, Jordan Green, Rus Heywood, Pan-Pan Jiang, Anton Kast, Marilyn Ladewig, Bob MacDonald, Philip Nelson, Katie Seaver, Joel Shor, Jimmy Tobin, Katrin Tomanek, and Subhashini Venugopalan. We gratefully acknowledge the support Project Euphonia received from members of the USM research team including Yu Zhang, Wei Han, Nanxin Chen, and many others. Most importantly, we wanted to say a huge thank you to the 2,200+ participants who recorded speech samples and the many advocacy groups who helped us connect with these participants.

1Audio volume has been adjusted for ease of listening, but the original files would be more consistent with those used in training and would have pauses, silences, variable volume, etc. ↩

Source: Google AI Blog


AVFormer: Injecting vision into frozen speech models for zero-shot AV-ASR

Posted by Arsha Nagrani and Paul Hongsuck Seo, Research Scientists, Google Research

Automatic speech recognition (ASR) is a well-established technology that is widely adopted for various applications such as conference calls, streamed video transcription and voice commands. While the challenges for this technology are centered around noisy audio inputs, the visual stream in multimodal videos (e.g., TV, online edited videos) can provide strong cues for improving the robustness of ASR systems — this is called audiovisual ASR (AV-ASR).

Although lip motion can provide strong signals for speech recognition and is the most common area of focus for AV-ASR, the mouth is often not directly visible in videos in the wild (e.g., due to egocentric viewpoints, face coverings, and low resolution) and therefore, a new emerging area of research is unconstrained AV-ASR (e.g., AVATAR), which investigates the contribution of entire visual frames, and not just the mouth region.

Building audiovisual datasets for training AV-ASR models, however, is challenging. Datasets such as How2 and VisSpeech have been created from instructional videos online, but they are small in size. In contrast, the models themselves are typically large and consist of both visual and audio encoders, and so they tend to overfit on these small datasets. Nonetheless, there have been a number of recently released large-scale audio-only models that are heavily optimized via large-scale training on massive audio-only data obtained from audio books, such as LibriLight and LibriSpeech. These models contain billions of parameters, are readily available, and show strong generalization across domains.

With the above challenges in mind, in “AVFormer: Injecting Vision into Frozen Speech Models for Zero-Shot AV-ASR”, we present a simple method for augmenting existing large-scale audio-only models with visual information, at the same time performing lightweight domain adaptation. AVFormer injects visual embeddings into a frozen ASR model (similar to how Flamingo injects visual information into large language models for vision-text tasks) using lightweight trainable adaptors that can be trained on a small amount of weakly labeled video data with minimum additional training time and parameters. We also introduce a simple curriculum scheme during training, which we show is crucial to enable the model to jointly process audio and visual information effectively. The resulting AVFormer model achieves state-of-the-art zero-shot performance on three different AV-ASR benchmarks (How2, VisSpeech and Ego4D), while also crucially preserving decent performance on traditional audio-only speech recognition benchmarks (i.e., LibriSpeech).

Unconstrained audiovisual speech recognition. We inject vision into a frozen speech model (BEST-RQ, in grey) for zero-shot audiovisual ASR via lightweight modules to create a parameter- and data-efficient model called AVFormer (blue). The visual context can provide helpful clues for robust speech recognition especially when the audio signal is noisy (the visual loaf of bread helps correct the audio-only mistake “clove” to “loaf” in the generated transcript).

Injecting vision using lightweight modules

Our goal is to add visual understanding capabilities to an existing audio-only ASR model while maintaining its generalization performance to various domains (both AV and audio-only domains).

To achieve this, we augment an existing state-of-the-art ASR model (Best-RQ) with the following two components: (i) linear visual projector and (ii) lightweight adapters. The former projects visual features in the audio token embedding space. This process allows the model to properly connect separately pre-trained visual feature and audio input token representations. The latter then minimally modifies the model to add understanding of multimodal inputs from videos. We then train these additional modules on unlabeled web videos from the HowTo100M dataset, along with the outputs of an ASR model as pseudo ground truth, while keeping the rest of the Best-RQ model frozen. Such lightweight modules enable data-efficiency and strong generalization of performance.

We evaluated our extended model on AV-ASR benchmarks in a zero-shot setting, where the model is never trained on a manually annotated AV-ASR dataset.


Curriculum learning for vision injection

After the initial evaluation, we discovered empirically that with a naïve single round of joint training, the model struggles to learn both the adapters and the visual projectors in one go. To mitigate this issue, we introduced a two-phase curriculum learning strategy that decouples these two factors — domain adaptation and visual feature integration — and trains the network in a sequential manner. In the first phase, the adapter parameters are optimized without feeding visual tokens at all. Once the adapters are trained, we add the visual tokens and train the visual projection layers alone in the second phase while the trained adapters are kept frozen.

The first stage focuses on audio domain adaptation. By the second phase, the adapters are completely frozen and the visual projector must simply learn to generate visual prompts that project the visual tokens into the audio space. In this way, our curriculum learning strategy allows the model to incorporate visual inputs as well as adapt to new audio domains in AV-ASR benchmarks. We apply each phase just once, as an iterative application of alternating phases leads to performance degradation.

Overall architecture and training procedure for AVFormer. The architecture consists of a frozen Conformer encoder-decoder model, and a frozen CLIP encoder (frozen layers shown in gray with a lock symbol), in conjunction with two lightweight trainable modules - (i) visual projection layer (orange) and bottleneck adapters (blue) to enable multimodal domain adaptation. We propose a two-phase curriculum learning strategy: the adapters (blue) are first trained without any visual tokens, after which the visual projection layer (orange) is tuned while all the other parts are kept frozen.

The plots below show that without curriculum learning, our AV-ASR model is worse than the audio-only baseline across all datasets, with the gap increasing as more visual tokens are added. In contrast, when the proposed two-phase curriculum is applied, our AV-ASR model performs significantly better than the baseline audio-only model.

Effects of curriculum learning. Red and blue lines are for audiovisual models and are shown on 3 datasets in the zero-shot setting (lower WER % is better). Using the curriculum helps on all 3 datasets (for How2 (a) and Ego4D (c) it is crucial for outperforming audio-only performance). Performance improves up until 4 visual tokens, at which point it saturates.

Results in zero-shot AV-ASR

We compare AVFormer to BEST-RQ, the audio version of our model, and AVATAR, the state of the art in AV-ASR, for zero-shot performance on the three AV-ASR benchmarks: How2, VisSpeech and Ego4D. AVFormer outperforms AVATAR and BEST-RQ on all, even outperforming both AVATAR and BEST-RQ when they are trained on LibriSpeech and the full set of HowTo100M. This is notable because for BEST-RQ, this involves training 600M parameters, while AVFormer only trains 4M parameters and therefore requires only a small fraction of the training dataset (5% of HowTo100M). Moreover, we also evaluate performance on LibriSpeech, which is audio-only, and AVFormer outperforms both baselines.

Comparison to state-of-the-art methods for zero-shot performance across different AV-ASR datasets. We also show performances on LibriSpeech which is audio-only. Results are reported as WER % (lower is better). AVATAR and BEST-RQ are finetuned end-to-end (all parameters) on HowTo100M whereas AVFormer works effectively even with 5% of the dataset thanks to the small set of finetuned parameters.

Conclusion

We introduce AVFormer, a lightweight method for adapting existing, frozen state-of-the-art ASR models for AV-ASR. Our approach is practical and efficient, and achieves impressive zero-shot performance. As ASR models get larger and larger, tuning the entire parameter set of pre-trained models becomes impractical (even more so for different domains). Our method seamlessly allows both domain transfer and visual input mixing in the same, parameter efficient model.


Acknowledgements

This research was conducted by Paul Hongsuck Seo, Arsha Nagrani and Cordelia Schmid.

Source: Google AI Blog


Identifying Disfluencies in Natural Speech

Posted by Dan Walker and Dan Liebling, Software Engineers, Google Research

People don’t write in the same way that they speak. Written language is controlled and deliberate, whereas transcripts of spontaneous speech (like interviews) are hard to read because speech is disorganized and less fluent. One aspect that makes speech transcripts particularly difficult to read is disfluency, which includes self-corrections, repetitions, and filled pauses (e.g., words like “umm”, and “you know”). Following is an example of a spoken sentence with disfluencies from the LDC CALLHOME corpus:

But that's it's not, it's not, it's, uh, it's a word play on what you just said.

It takes some time to understand this sentence — the listener must filter out the extraneous words and resolve all of the nots. Removing the disfluencies makes the sentence much easier to read and understand:

But it’s a word play on what you just said.

While people generally don't even notice disfluencies in day-to-day conversation, early foundational work in computational linguistics demonstrated how common they are. In 1994, using the Switchboard corpus, Elizabeh Shriberg demonstrated that there is a 50% probability for a sentence of 10–13 words to include a disfluency and that the probability increases with sentence length.

The proportion of sentences from the Switchboard dataset with at least one disfluency plotted against sentence length measured in non-disfluent (i.e., efficient) tokens in the sentence. The longer a sentence gets, the more likely it is to contain a disfluency.

In “Teaching BERT to Wait: Balancing Accuracy and Latency for Streaming Disfluency Detection”, we present research findings on how to “clean up” transcripts of spoken text. We create more readable transcripts and captions of human speech by finding and removing disfluencies in people’s speech. Using labeled data, we created machine learning (ML) algorithms that identify disfluencies in human speech. Once those are identified we can remove the extra words to make transcripts more readable. This also improves the performance of natural language processing (NLP) algorithms that work on transcripts of human speech. Our work puts special priority on ensuring that these models are able to run on mobile devices so that we can protect user privacy and preserve performance in scenarios with low connectivity.

Base Model Overview
At the core of our base model is a pre-trained BERTBASE encoder with 108.9 million parameters. We use the standard per-token classifier configuration, with a binary classification head being fed by the sequence encodings for each token.

Illustration of how tokens in text become numerical embeddings, which then lead to output labels.

We refined the BERT encoder by continuing the pretraining on the comments from the Pushrift Reddit dataset from 2019. Reddit comments are not speech data, but are more informal and conversational than the wiki and book data. This trains the encoder to better understand informal language, but may run the risk of internalizing some of the biases inherent in the data. For our particular use case, however, the model only captures the syntax or overall form of the text, not its content, which avoids potential issues related to semantic-level biases in the data.

We fine-tune our model for disfluency classification on hand-labeled corpora, such as the Switchboard corpus mentioned above. Hyperparameters (batch size, learning rate, number of training epochs, etc.) were optimized using Vizier.

We also produce a range of “small” models for use on mobile devices using a knowledge distillation technique known as “self training”. Our best small model is based on the Small-vocab BERT variant with 3.1 million parameters. This smaller model achieves comparable results to our baseline at 1% the size (in MiB). You can read more about how we achieved this model miniaturization in our 2021 Interspeech paper.

Streaming
Some of the latest use cases for automatic speech transcription include automated live captioning, such as produced by the Android “Live Captions” feature, which automatically transcribes spoken language in audio being played on the device. For disfluency removal to be of use in improving the readability of the captions in this setting, then it must happen quickly and in a stable manner. That is, the model should not change its past predictions as it sees new words in the transcript.

We call this live token-by-token processing streaming. Accurate streaming is difficult because of temporal dependencies; most disfluencies are only recognizable later. For example, a repetition does not actually become a repetition until the second time the word or phrase is said.

To investigate whether our disfluency detection model is effective in streaming applications, we split the utterances in our training set into prefix segments, where only the first N tokens of the utterance were provided at training time, for all values of N up to the full length of the utterance. We evaluated the model simulating a stream of spoken text by feeding prefixes to the models and measuring the performance with several metrics that capture model accuracy, stability, and latency including streaming F1, time to detection (TTD), edit overhead (EO), and average wait time (AWT). We experimented with look-ahead windows of either one or two tokens, allowing the model to “peek” ahead at additional tokens for which the model is not required to produce a prediction. In essence, we’re asking the model to “wait” for one or two more tokens of evidence before making a decision.

While adding this fixed look-ahead did improve the stability and streaming F1 scores in many contexts, we found that in some cases the label was already clear even without looking ahead to the next token and the model did not necessarily benefit from waiting. Other times, waiting for just one extra token was sufficient. We hypothesized that the model itself could learn when it should wait for more context. Our solution was a modified model architecture that includes a “wait” classification head that decides when the model has seen enough evidence to trust the disfluency classification head.

Diagram showing how the model labels input tokens as they arrive. The BERT embedding layers feed into two separate classification heads, which are combined for the output.

We constructed a training loss function that is a weighted sum of three factors:

  1. The traditional cross-entropy loss for the disfluency classification head
  2. A cross-entropy term that only considers up to the first token with a “wait” classification
  3. A latency penalty that discourages the model from waiting too long to make a prediction

We evaluated this streaming model as well as the standard baseline with no look-ahead and with both 1- and 2-token look-ahead values:

Graph of the streaming F1 score versus the average wait time in tokens. Three data points indicate F1 scores above 0.82 across multiple wait times. The proposed streaming model achieves near top performance with much shorter wait times than the fixed look ahead models.

The streaming model achieved a better streaming F1 score than both a standard baseline with no look ahead and a model with a look ahead of 1. It performed nearly as well as the variant with fixed look ahead of 2, but with much less waiting. On average the model waited for only 0.21 tokens of context.

Internationalization
Our best outcomes so far have been with English transcripts. This is mostly due to resourcing issues: while there are a number of relatively large labeled conversational datasets that include disfluencies in English, other languages often have very few such datasets available. So, in order to make disfluency detection models available outside English a method is needed to build models in a way that does not require finding and labeling hundreds of thousands of utterances in each target language. A promising solution is to leverage multi-language versions of BERT to transfer what a model has learned about English disfluencies to other languages in order to achieve similar performance with much less data. This is an area of active research, but we do have some promising results to outline here.

As a first effort to validate this approach, we added labels to about 10,000 lines of dialogue from the German CALLHOME dataset. We then started with the Geotrend English and German Bilingual BERT model (extracted from Multilingual BERT) and fine-tuned it with approximately 77,000 disfluency-labeled English Switchboard examples and 1.3 million examples of self-labeled transcripts from the Fisher Corpus. Then, we did further fine tuning with about 7,500 in-house–labeled examples from the German CALLHOME dataset.

Diagram illustrating the flow of labeled data and self-trained output in our best multilingual training setup. By training on both English and German data we are able to improve performance via transfer learning.

Our results indicate that fine-tuning on a large English corpus can produce acceptable precision using zero-shot transfer to similar languages like German, but at least a modest amount of German labels were needed to improve recall from less than 60% to greater than 80%. Two-stage fine-tuning of an English-German bilingual model produced the highest precision and overall F1 score.

Approach Precision Recall F1
German BERTBASE model fine-tuned on 7,300 human-labeled German CALLHOME examples 89.1% 81.3% 85.0
Same as above but with additional 7,500 self-labeled German CALLHOME examples 91.5% 83.3% 87.2
English/German Bilingual BERTbase model fine-tuned on English Switchboard+Fisher, evaluated on German CALLHOME (zero-shot language transfer) 87.2% 59.1% 70.4
Same as above but subsequently fine-tuned with 14,800 German CALLHOME (human- and self-labeled) examples 95.5% 82.6% 88.6

Conclusion
Cleaning up disfluencies from transcripts can improve not just their readability for people, but also the performance of other models that consume transcripts. We demonstrate effective methods for identifying disfluencies and expand our disfluency model to resource-constrained environments, new languages, and more interactive use cases.

Acknowledgements
Thank you to Vicky Zayats, Johann Rocholl, Angelica Chen, Noah Murad, Dirk Padfield, and Preeti Mohan for writing the code, running the experiments, and composing the papers discussed here. Wealso thank our technical product manager Aaron Schneider, Bobby Tran from the Cerebra Data Ops team, and Chetan Gupta from Speech Data Ops for their support obtaining additional data labels.

Source: Google AI Blog


Personalized ASR Models from a Large and Diverse Disordered Speech Dataset

Posted by Katrin Tomanek, Software Engineer and Bob MacDonald, Technical Program Manager, Google Research

Speech impairments affect millions of people, with underlying causes ranging from neurological or genetic conditions to physical impairment, brain damage or hearing loss. Similarly, the resulting speech patterns are diverse, including stuttering, dysarthria, apraxia, etc., and can have a detrimental impact on self-expression, participation in society and access to voice-enabled technologies. Automatic speech recognition (ASR) technologies have the potential to help individuals with such speech impairments by improving access to dictation and home automation and by enhancing communication. However, while the increased computational power of deep learning systems and the availability of large training datasets has improved the accuracy of ASR systems, their performance is still insufficient for many people with speech disorders, rendering the technology unusable for many of the speakers who could benefit the most.

In 2019, we introduced Project Euphonia and discussed how we could use personalized ASR models of disordered speech to achieve accuracies on par with non-personalized ASR on typical speech. Today we share the results of two studies, presented at Interspeech 2021, that aim to expand the availability of personalized ASR models to more users. In “Disordered Speech Data Collection: Lessons Learned at 1 Million Utterances from Project Euphonia”, we present a greatly expanded collection of disordered speech data, composed of over 1 million utterances. Then, in “Automatic Speech Recognition of Disordered Speech: Personalized models outperforming human listeners on short phrases”, we discuss our efforts to generate personalized ASR models based on this corpus. This approach leads to highly accurate models that can achieve up to 85% improvement to the word error rate in select domains compared to out-of-the-box speech models trained on typical speech.

Impaired Speech Data Collection
Since 2019, speakers with speech impairments of varying degrees of severity across a variety of conditions have provided voice samples to support Project Euphonia’s research mission. This effort has grown Euphonia’s corpus to over 1 million utterances, comprising over 1400 hours from 1330 speakers (as of August 2021).

Distribution of severity of speech disorder and condition across all speakers with more than 300 utterances recorded. For conditions, only those with > 5 speakers are shown (all others aggregated into “OTHER” for k-anonymity).
ALS = amyotrophic lateral sclerosis; DS = Down syndrome; PD = Parkinson’s disease; CP = cerebral palsy; HI = hearing impaired; MD = muscular dystrophy; MS = multiple sclerosis

To simplify the data collection, participants used an at-home recording system on their personal hardware (laptop or phone, with and without headphones), instead of an idealized lab-based setting that would collect studio quality recordings.

To reduce transcription cost, while still maintaining high transcript conformity, we prioritized scripted speech. Participants read prompts shown on a browser-based recording tool. Phrase prompts covered use-cases like home automation (“Turn on the TV.”), caregiver conversations (“I am hungry.”) and informal conversations (“How are you doing? Did you have a nice day?”). Most participants received a list of 1500 phrases, which included 1100 unique phrases along with 100 phrases that were each repeated four more times.

Speech professionals conducted a comprehensive auditory-perceptual speech assessment while listening to a subset of utterances for every speaker providing the following speaker-level metadata: speech disorder type (e.g., stuttering, dysarthria, apraxia), rating of 24 features of abnormal speech (e.g., hypernasality, articulatory imprecision, dysprosody), as well as recording quality assessments of both technical (e.g., signal dropouts, segmentation problems) and acoustic (e.g., environmental noise, secondary speaker crosstalk) features.

Personalized ASR Models
This expanded impaired speech dataset is the foundation of our new approach to personalized ASR models for disordered speech. Each personalized model uses a standard end-to-end, RNN-Transducer (RNN-T) ASR model that is fine-tuned using data from the target speaker only.

Architecture of RNN-Transducer. In our case, the encoder network consists of 8 layers and the predictor network consists of 2 layers of uni-directional LSTM cells.

To accomplish this, we focus on adapting the encoder network, i.e. the part of the model dealing with the specific acoustics of a given speaker, as speech sound disorders were most common in our corpus. We found that only updating the bottom five (out of eight) encoder layers while freezing the top three encoder layers (as well as the joint layer and decoder layers) led to the best results and effectively avoided overfitting. To make these models more robust against background noise and other acoustic effects, we employ a configuration of SpecAugment specifically tuned to the prevailing characteristics of disordered speech. Further, we found that the choice of the pre-trained base model was critical. A base model trained on a large and diverse corpus of typical speech (multiple domains and acoustic conditions) proved to work best for our scenario.

Results
We trained personalized ASR models for ~430 speakers who recorded at least 300 utterances. 10% of utterances were held out as a test set (with no phrase overlap) on which we calculated the word error rate (WER) for the personalized model and the unadapted base model.

Overall, our personalization approach yields significant improvements across all severity levels and conditions. Even for severely impaired speech, the median WER for short phrases from the home automation domain dropped from around 89% to 13%. Substantial accuracy improvements were also seen across other domains such as conversational and caregiver.

WER of unadapted and personalized ASR models on home automation phrases.

To understand when personalization does not work well, we analyzed several subgroups:

  • HighWER and LowWER: Speakers with high and low personalized model WERs based on the 1st and 5th quintiles of the WER distribution.
  • SurpHighWER: Speakers with a surprisingly high WER (participants with typical speech or mild speech impairment of the HighWER group).

Different pathologies and speech disorder presentations are expected to impact ASR non-uniformly. The distribution of speech disorder types within the HighWER group indicates that dysarthria due to cerebral palsy was particularly difficult to model. Not surprisingly, median severity was also higher in this group.

To identify the speaker-specific and technical factors that impact ASR accuracy, we examined the differences (Cohen's D) in the metadata between the participants that had poor (HighWER) and excellent (LowWER) ASR performance. As expected, overall speech severity was significantly lower in the LowWER group than in the HighWER group (p < 0.01). Intelligibility and severity were the most prominent atypical speech features in the HighWER group; however, other speech features also emerged, including abnormal prosody, articulation, and phonation. These speech features are known to degrade overall speech intelligibility.

The SurpHighWER group had fewer training utterances and lower SNR compared with the LowWER group (p < 0.01) resulting in large (negative) effect sizes, with all other factors having small effect sizes, except fastness. In contrast, the HighWER group exhibited medium to large differences across all factors.

Speech disorder and technical metadata effect sizes for the HighWER-vs-LowWER and SurpHighWER-vs-LowWER pairs. Positive effects indicated that the group values of the HighWER group were greater than LowWER groups.

We then compared personalized ASR models to human listeners. Three speech professionals independently transcribed 30 utterances per speaker. We found that WERs were, on average, lower for personalized ASR models compared to the WERs of human listeners, with gains increasing by severity.

Delta between the WERs of the personalized ASR models and the human listeners. Negative values indicate that personalized ASR performs better than human (expert) listeners.

Conclusions
With over 1 million utterances, Euphonia’s corpus is one of the largest and most diversely disordered speech corpora (in terms of disorder types and severities) and has enabled significant advances in ASR accuracy for these types of atypical speech. Our results demonstrate the efficacy of personalized ASR models for recognizing a wide range of speech impairments and severities, with potential for making ASR available to a wider population of users.

Acknowledgements
Key contributors to this project include Michael Brenner, Julie Cattiau, Richard Cave, Jordan Green, Rus Heywood, Pan-Pan Jiang, Anton Kast, Marilyn Ladewig, Bob MacDonald, Phil Nelson, Katie Seaver, Jimmy Tobin, and Katrin Tomanek. We gratefully acknowledge the support Project Euphonia received from members of many speech research teams across Google, including Françoise Beaufays, Fadi Biadsy, Dotan Emanuel, Khe Chai Sim, Pedro Moreno Mengibar, Arun Narayanan, Hasim Sak, Suzan Schwartz, Joel Shor, and many others. And most importantly, we wanted to say a huge thank you to the over 1300 participants who recorded speech samples and the many advocacy groups who helped us connect with these participants.

Source: Google AI Blog


On-Device Captioning with Live Caption

Posted by Michelle Tadmor-Ramanovich and Nadav Bar, Senior Software Engineers, Google Research, Tel-Aviv

Captions for audio content are essential for the deaf and hard of hearing, but they benefit everyone. Watching video without audio is common — whether on the train, in meetings, in bed or when the kids are asleep — and studies have shown that subtitles can increase the duration of time that users spend watching a video by almost 40%. Yet caption support is fragmented across apps and even within them, resulting in a significant amount of audio content that remains inaccessible, including live blogs, podcasts, personal videos, audio messages, social media and others.
Recently we introduced Live Caption, a new Android feature that automatically captions media playing on your phone. The captioning happens in real time, completely on-device, without using network resources, thus preserving privacy and lowering latency. The feature is currently available on Pixel 4 and Pixel 4 XL, will roll out to Pixel 3 models later this year, and will be more widely available on other Android devices soon.
When media is playing, Live Caption can be launched with a single tap from the volume control to display a caption box on the screen.
Building Live Caption for Accuracy and Efficiency
Live Caption works through a combination of three on-device deep learning models: a recurrent neural network (RNN) sequence transduction model for speech recognition (RNN-T), a text-based recurrent neural network model for unspoken punctuation, and a convolutional neural network (CNN) model for sound events classification. Live Caption integrates the signal from the three models to create a single caption track, where sound event tags, like [APPLAUSE] and [MUSIC], appear without interrupting the flow of speech recognition results. Punctuation symbols are predicted while text is updated in parallel.

Incoming sound is processed through a Sound Recognition and ASR feedback loop. The produced text or sound label is formatted and added to the caption.
For sound recognition, we leverage previous work that was done for sound events detection, using a model that was built on top of the AudioSet dataset. The Sound Recognition model is used not only to generate popular sound effect labels but also to detect speech periods. The full automatic speech recognition (ASR) RNN-T engine runs only during speech periods in order to minimize memory and battery usage. For example, when music is detected and speech is not present in the audio stream, the [MUSIC] label will appear on screen, and the ASR model will be unloaded. The ASR model is only loaded back into memory when speech is present in the audio stream again.

In order for Live Caption to be most useful, it should be able to run continuously for long periods of time. To do this, Live Caption’s ASR model is optimized for edge-devices using several techniques, such as neural connection pruning, which reduced the power consumption to 50% compared to the full sized speech model. Yet while the model is significantly more energy efficient, it still performs well for a variety of use cases, including captioning videos, recognizing short queries and narrowband telephony speech, while also being robust to background noise.

The text-based punctuation model was optimized for running continuously on-device using a smaller architecture than the cloud equivalent, and then quantized and serialized using the TensorFlow Lite runtime. As the caption is formed, speech recognition results are rapidly updated a few times per second. In order to save on computational resources and provide a smooth user experience, the punctuation prediction is performed on the tail of the text from the most recently recognized sentence, and if the next updated ASR results do not change that text, the previously punctuated results are retained and reused.

Looking forward
Live Caption is now available in English on Pixel 4 and will soon be available on Pixel 3 and other Android devices. We look forward to bringing this feature to more users by expanding its support to other languages and by further improving the formatting in order to improve the perceived accuracy and coherency of the captions, particularly for multi-speaker content.

Acknowledgements
The core team includes Robert Berry, Anthony Tripaldi, Danielle Cohen, Anna Belozovsky, Yoni Tsafir, Elliott Burford, Justin Lee, Kelsie Van Deman, Nicole Bleuel, Brian Kemler, and Benny Schlesinger. We would like to thank the Google Speech team, especially Qiao Liang, Arun Narayanan, and Rohit Prabhavalkar for their insightful work on the ASR model as well as Chung-Cheng Chiu from Google Brain Team; Dan Ellis and Justin Paul for their help with integrating the Sound Recognition model; Tal Remez for his help in developing the punctuation model; Kevin Rocard and Eric Laurent‎ for their help with the Android audio capture API; and Eugenio Marchiori, Shivanker Goel, Ye Wen, Jay Yoo, Asela Gunawardana, and Tom Hume for their help with the Android infrastructure work.

Source: Google AI Blog


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

Posted by Arindrima Datta and Anjuli Kannan, Software Engineers, Google Research

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


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

Posted by Joel Shor and Dotan Emanuel, Research Engineers, Google Research, Tel Aviv

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


SpecAugment: A New Data Augmentation Method for Automatic Speech Recognition

Posted by Daniel S. Park, AI Resident and William Chan, Research Scientist

Automatic Speech Recognition (ASR), the process of taking an audio input and transcribing it to text, has benefited greatly from the ongoing development of deep neural networks. As a result, ASR has become ubiquitous in many modern devices and products, such as Google Assistant, Google Home and YouTube. Nevertheless, there remain many important challenges in developing deep learning-based ASR systems. One such challenge is that ASR models, which have many parameters, tend to overfit the training data and have a hard time generalizing to unseen data when the training set is not extensive enough.

In the absence of an adequate volume of training data, it is possible to increase the effective size of existing data through the process of data augmentation, which has contributed to significantly improving the performance of deep networks in the domain of image classification. In the case of speech recognition, augmentation traditionally involves deforming the audio waveform used for training in some fashion (e.g., by speeding it up or slowing it down), or adding background noise. This has the effect of making the dataset effectively larger, as multiple augmented versions of a single input is fed into the network over the course of training, and also helps the network become robust by forcing it to learn relevant features. However, existing conventional methods of augmenting audio input introduces additional computational cost and sometimes requires additional data.

In our recent paper, “SpecAugment: A Simple Data Augmentation Method for Automatic Speech Recognition”, we take a new approach to augmenting audio data, treating it as a visual problem rather than an audio one. Instead of augmenting the input audio waveform as is traditionally done, SpecAugment applies an augmentation policy directly to the audio spectrogram (i.e., an image representation of the waveform). This method is simple, computationally cheap to apply, and does not require additional data. It is also surprisingly effective in improving the performance of ASR networks, demonstrating state-of-the-art performance on the ASR tasks LibriSpeech 960h and Switchboard 300h.

SpecAugment
In traditional ASR, the audio waveform is typically encoded as a visual representation, such as a spectrogram, before being input as training data for the network. Augmentation of training data is normally applied to the waveform audio before it is converted into the spectrogram, such that after every iteration, new spectrograms must be generated. In our approach, we investigate the approach of augmenting the spectrogram itself, rather than the waveform data. Since the augmentation is applied directly to the input features of the network, it can be run online during training without significantly impacting training speed.
A waveform is typically converted into a visual representation (in our case, a log mel spectrogram; steps 1 through 3 of this article) before being fed into a network.
SpecAugment modifies the spectrogram by warping it in the time direction, masking blocks of consecutive frequency channels, and masking blocks of utterances in time. These augmentations have been chosen to help the network to be robust against deformations in the time direction, partial loss of frequency information and partial loss of small segments of speech of the input. An example of such an augmentation policy is displayed below.
The log mel spectrogram is augmented by warping in the time direction, and masking (multiple) blocks of consecutive time steps (vertical masks) and mel frequency channels (horizontal masks). The masked portion of the spectrogram is displayed in purple for emphasis.
To test SpecAugment, we performed some experiments with the LibriSpeech dataset, where we took three Listen Attend and Spell (LAS) networks, end-to-end networks commonly used for speech recognition, and compared the test performance between networks trained with and without augmentation. The performance of an ASR network is measured by the Word Error Rate (WER) of the transcript produced by the network against the target transcript. Here, all hyperparameters were kept the same, and only the data fed into the network was altered. We found that SpecAugment improves network performance without any additional adjustments to the network or training parameters.
Performance of networks on the test sets of LibriSpeech with and without augmentation. The LibriSpeech test set is divided into two portions, test-clean and test-other, the latter of which contains noisier audio data.
More importantly, SpecAugment prevents the network from over-fitting by giving it deliberately corrupted data. As an example of this, below we show how the WER for the training set and the development (or dev) set evolves through training with and without augmentation. We see that without augmentation, the network achieves near-perfect performance on the training set, while grossly under-performing on both the clean and noisy dev set. On the other hand, with augmentation, the network struggles to perform as well on the training set, but actually shows better performance on the clean dev set, and shows comparable performance on the noisy dev set. This suggests that the network is no longer over-fitting the training data, and that improving training performance would lead to better test performance.
Training, clean (dev-clean) and noisy (dev-other) development set performance with and without augmentation.
State-of-the-Art Results
We can now focus on improving training performance, which can be done by adding more capacity to the networks by making them larger. By doing this in conjunction with increasing training time, we were able to get state-of-the-art (SOTA) results on the tasks LibriSpeech 960h and Switchboard 300h.
Word error rates (%) for state-of-the-art results for the tasks LibriSpeech 960h and Switchboard 300h. The test set for both tasks have a clean (clean/Switchboard) and a noisy (other/CallHome) subset. Previous SOTA results taken from Li et. al (2019), Yang et. al (2018) and Zeyer et. al (2018).
The simple augmentation scheme we have used is surprisingly powerful—we are able to improve the performance of the end-to-end LAS networks so much that it surpasses those of classical ASR models, which traditionally did much better on smaller academic datasets such as LibriSpeech or Switchboard.
Performance of various classes of networks on LibriSpeech and Switchboard tasks. The performance of LAS models is compared to classical (e.g., HMM) and other end-to-end models (e.g., CTC/ASG) over time.
Language Models
Language models (LMs), which are trained on a bigger corpus of text-only data, have played a significant role in improving the performance of an ASR network by leveraging information learned from text. However, LMs typically need to be trained separately from the ASR network, and can be very large in memory, making it hard to fit on a small device, such as a phone. An unexpected outcome of our research was that models trained with SpecAugment out-performed all prior methods even without the aid of a language model. While our networks still benefit from adding an LM, our results are encouraging in that it suggests the possibility of training networks that can be used for practical purposes without the aid of an LM.
Word error rates for LibriSpeech and Switchboard tasks with and without LMs. SpecAugment outperforms previous state-of-the-art even before the inclusion of a language model.
Most of the work on ASR in the past has been focused on looking for better networks to train. Our work demonstrates that looking for better ways to train networks is a promising alternative direction of research.

Acknowledgements
We would like to thank the co-authors of our paper Chung-Cheng Chiu, Ekin Dogus Cubuk, Quoc Le, Yu Zhang and Barret Zoph. We also thank Yuan Cao, Ciprian Chelba, Kazuki Irie, Ye Jia, Anjuli Kannan, Patrick Nguyen, Vijay Peddinti, Rohit Prabhavalkar, Yonghui Wu and Shuyuan Zhang for useful discussions.

Source: Google AI Blog


Real-time Continuous Transcription with Live Transcribe

Posted by Sagar Savla, Product Manager, Machine Perception

The World Health Organization (WHO) estimates that there are 466 million people globally that are deaf and hard of hearing. A crucial technology in empowering communication and inclusive access to the world's information to this population is automatic speech recognition (ASR), which enables computers to detect audible languages and transcribe them into text for reading. Google's ASR is behind automated captions in Youtube, presentations in Slides and also phone calls. However, while ASR has seen multiple improvements in the past couple of years, the deaf and hard of hearing still mainly rely on manual-transcription services like CART in the US, Palantypist in the UK, or STTR in other countries. These services can be prohibitively expensive and often require to be scheduled far in advance, diminishing the opportunities for the deaf and hard of hearing to participate in impromptu conversations as well as social occasions. We believe that technology can bridge this gap and empower this community.

Today, we're announcing Live Transcribe, a free Android service that makes real-world conversations more accessible by bringing the power of automatic captioning into everyday, conversational use. Powered by Google Cloud, Live Transcribe captions conversations in real-time, supporting over 70 languages and more than 80% of the world's population. You can launch it with a single tap from within any app, directly from the accessibility icon on the system tray.

Building Live Transcribe
Previous ASR-based transcription systems have generally required compute-intensive models, exhaustive user research and expensive access to connectivity, all which hinder the adoption of automated continuous transcription. To address these issues and ensure reasonably accurate real-time transcription, Live Transcribe combines the results of extensive user experience (UX) research with seamless and sustainable connectivity to speech processing servers. Furthermore, we needed to ensure that connectivity to these servers didn't cause our users excessive data usage.

Relying on cloud ASR provides us greater accuracy, but we wanted to reduce the network data consumption that Live Transcribe requires. To do this, we implemented an on-device neural network-based speech detector, built on our previous work with AudioSet. This network is an image-like model, similar to our published VGGish model, which detects speech and automatically manages network connections to the cloud ASR engine, minimizing data usage over long periods of use.

User Experience
To make Live Transcribe as intuitive as possible, we partnered with Gallaudet University to kickstart user experience research collaborations that would ensure core user needs were satisfied while maximizing the potential of our technologies. We considered several different modalities, computers, tablets, smartphones, and even small projectors, iterating ways to display auditory information and captions. In the end, we decided to focus on the smartphone form factor because of the sheer ubiquity of these devices and the increasing capabilities they have.

Once this was established, we needed to address another important issue: displaying transcription confidence. Traditionally considered to be helpful to the user, our research explored whether we actually needed to show word-level or phrase-level confidence.
Displaying confidence level of the transcription. Yellow is high confidence, green is medium and blue is low confidence. White is fresh text awaiting context before finalizing. On the left, the coloring is at a per-phrase level while on the right is at a per-word level.1 Research found them to be distracting to the user without providing conversational value.
Reinforcing previous UX research in this space, our research shows that a transcript is easiest to read when it is not layered with these signals. Instead, Live Transcribe focuses on better presentation of the text and supplementing it with other auditory signals besides speech.

Another useful UX signal is the noise level of their current environment. Known as the cocktail party problem, understanding a speaker in a noisy room is a major challenge for computers. To address this, we built an indicator that visualizes the volume of user speech relative to background noise. This also gives users instant feedback on how well the microphone is receiving the incoming speech from the speaker, allowing them to adjust the placement of the phone.
The loudness and noise indicator is made of two concentric circles. The inner brighter circle, indicating the noise floor, tells a deaf user how audibly noisy the current environment is. The outer circle shows how well the speaker’s voice is received.Together, the circles visually show the relative difference intuitively.
Future Work
Potential future improvements in mobile-based automatic speech transcription include on-device recognition, speaker-separation, and speech enhancement. Relying solely on transcription can have pitfalls that can lead to miscommunication. Our research with Gallaudet University shows that combining it with other auditory signals like speech detection and a loudness indicator, makes a tangibly meaningful change in communication options for our users.

Live Transcribe is now available in a staged rollout on the Play Store, and is pre-installed on all Pixel 3 devices with the latest update. Live Transcribe can then be enabled via the Accessibility Settings. You can also read more about it on The Keyword.

Acknowledgements
Live Transcribe was made by researchers Chet Gnegy, Dimitri Kanevsky, and Justin S. Paul in collaboration with Android Accessibility team members Brian Kemler, Thomas Lin, Alex Huang, Jacqueline Huang, Ben Chung, Richard Chang, I-ting Huang, Jessie Lin, Ausmus Chang, Weiwei Wei, Melissa Barnhart and Bingying Xia. We'd also like to thank our close partners from Gallaudet University, Christian Vogler, Norman Williams and Paula Tucker.

1 Eagle-eyed readers can see the phrase level confidence mode in use by Dr. Obeidat in the video above.↩


Source: Google AI Blog


Kaldi now offers TensorFlow integration

Posted by Raziel Alvarez, Staff Research Engineer at Google and Yishay Carmiel, Founder of IntelligentWire

Automatic speech recognition (ASR) has seen widespread adoption due to the recent proliferation of virtual personal assistants and advances in word recognition accuracy from the application of deep learning algorithms. Many speech recognition teams rely on Kaldi, a popular open-source speech recognition toolkit. We're announcing today that Kaldi now offers TensorFlow integration.

With this integration, speech recognition researchers and developers using Kaldi will be able to use TensorFlow to explore and deploy deep learning models in their Kaldi speech recognition pipelines. This will allow the Kaldi community to build even better and more powerful ASR systems as well as providing TensorFlow users with a path to explore ASR while drawing upon the experience of the large community of Kaldi developers.

Building an ASR system that can understand human speech in every language, accent, environment, and type of conversation is an extremely complex undertaking. A traditional ASR system can be seen as a processing pipeline with many separate modules, where each module operates on the output from the previous one. Raw audio data enters the pipeline at one end and a transcription of recognized speech emerges from the other. In the case of Kaldi, these ASR transcriptions are post processed in a variety of ways to support an increasing array of end-user applications.

Yishay Carmiel and Hainan Xu of Seattle-based IntelligentWire, who led the development of the integration between Kaldi and TensorFlow with support from the two teams, know this complexity first-hand. Their company has developed cloud software to bridge the gap between live phone conversations and business applications. Their goal is to let businesses analyze and act on the contents of the thousands of conversations their representatives have with customers in real-time and automatically handle tasks like data entry or responding to requests. IntelligentWire is currently focused on the contact center market, in which more than 22 million agents throughout the world spend 50 billion hours a year on the phone and about 25 billion hours interfacing with and operating various business applications.

For an ASR system to be useful in this context, it must not only deliver an accurate transcription but do so with very low latency in a way that can be scaled to support many thousands of concurrent conversations efficiently. In situations like this, recent advances in deep learning can help push technical limits, and TensorFlow can be very useful.

In the last few years, deep neural networks have been used to replace many existing ASR modules, resulting in significant gains in word recognition accuracy. These deep learning models typically require processing vast amounts of data at scale, which TensorFlow simplifies. However, several major challenges must still be overcome when developing production-grade ASR systems:

  • Algorithms - Deep learning algorithms give the best results when tailored to the task at hand, including the acoustic environment (e.g. noise), the specific language spoken, the range of vocabulary, etc. These algorithms are not always easy to adapt once deployed.
  • Data - Building an ASR system for different languages and different acoustic environments requires large quantities of multiple types of data. Such data may not always be available or may not be suitable for the use case.
  • Scale - ASR systems that can support massive amounts of usage and many languages typically consume large amounts of computational power.

One of the ASR system modules that exemplifies these challenges is the language model. Language models are a key part of most state-of-the-art ASR systems; they provide linguistic context that helps predict the proper sequence of words and distinguish between words that sound similar. With recent machine learning breakthroughs, speech recognition developers are now using language models based on deep learning, known as neural language models. In particular, recurrent neural language models have shown superior results over classic statistical approaches.

However, the training and deployment of neural language models is complicated and highly time-consuming. For IntelligentWire, the integration of TensorFlow into Kaldi has reduced the ASR development cycle by an order of magnitude. If a language model already exists in TensorFlow, then going from model to proof of concept can take days rather than weeks; for new models, the development time can be reduced from months to weeks. Deploying new TensorFlow models into production Kaldi pipelines is straightforward as well, providing big gains for anyone working directly with Kaldi as well as the promise of more intelligent ASR systems for everyone in the future.

Similarly, this integration provides TensorFlow developers with easy access to a robust ASR platform and the ability to incorporate existing speech processing pipelines, such as Kaldi's powerful acoustic model, into their machine learning applications. Kaldi modules that feed the training of a TensorFlow deep learning model can be swapped cleanly, facilitating exploration, and the same pipeline that is used in production can be reused to evaluate the quality of the model.

We hope this Kaldi-TensorFlow integration will bring these two vibrant open-source communities closer together and support a wide variety of new speech-based products and related research breakthroughs. To get started using Kaldi with TensorFlow, please check out the Kaldi repo and also take a look at an example for Kaldi setup running with TensorFlow.