Month: July 2021

Posted by Samuel J. Yang, Research Scientist and Dick Lyon, Principal Scientist, Google Research

For the ~466 million people in the world who are deaf or hard of hearing, the lack of easy access to accessibility services can be a barrier to participating in spoken conversations encountered daily. While hearing aids can help alleviate this, simply amplifying sound is insufficient for many. One additional option that may be available is the cochlear implant (CI), which is an electronic device that is surgically inserted into a part of the inner ear, called the cochlea, and stimulates the auditory nerve electrically via external sound processors. While many individuals with these cochlear implants can learn to interpret these electrical stimulations as audible speech, the listening experience can be quite varied and particularly challenging in noisy environments.

Modern cochlear implants drive electrodes with pulsatile signals (i.e., discrete stimulation pulses) that are computed by external sound processors. The main challenge still facing the CI field is how to best process sounds — to convert sounds to pulses on electrodes — in a way that makes them more intelligible to users. Recently, to stimulate progress on this problem, scientists in industry and academia organized a CI Hackathon to open the problem up to a wider range of ideas.

In this post, we share exploratory research demonstrating that a speech enhancement preprocessor — specifically, a noise suppressor — can be used at the input of a CI’s processor to enhance users’ understanding of speech in noisy environments. We also discuss how we built on this work in our entry for the CI Hackathon and how we will continue developing this work.

Improving CIs with Noise Suppression
In 2019, a small internal project demonstrated the benefits of noise suppression at the input of a CI’s processor. In this project, participants listened to 60 pre-recorded and pre-processed audio samples and ranked them by their listening comfort. CI users listened to the audio using their devices’ existing strategy for generating electrical pulses.

As shown below, both listening comfort and intelligibility usually increased, sometimes dramatically, when speech with noise (the lightest bar) was processed with noise suppression.

CI users in an early research study have improved listening comfort — qualitatively scored from “very poor” (0.0) to “OK” (0.5) to “very good” (1.0) — and speech intelligibility (i.e., the fraction of words in a sentence correctly transcribed) when trying to listen to noisy audio samples of speech with noise suppression applied.

For the CI Hackathon, we built on the project above, continuing to leverage our use of a noise suppressor while additionally exploring an approach to compute the pulses too

Overview of the Processing Approach
The hackathon considered a CI with 16 electrodes. Our approach decomposes the audio into 16 overlapping frequency bands, corresponding to the positions of the electrodes in the cochlea. Next, because the dynamic range of sound easily spans multiple orders of magnitude more than what we expect the electrodes to represent, we aggressively compress the dynamic range of the signal by applying “per-channel energy normalization” (PCEN). Finally, the range-compressed signals are used to create the electrodogram (i.e., what the CI displays on the electrodes).

In addition, the hackathon required a submission be evaluated in multiple audio categories, including music, which is an important but notoriously difficult category of sounds for CI users to enjoy. However, the speech enhancement network was trained to suppress non-speech sounds, including both noise and music, so we needed to take extra measures to avoid suppressing instrumental music (note that in general, music suppression might be preferred by some users in certain contexts). To do this, we created a “mix” of the original audio with the noise-suppressed audio so that enough of the music would pass through to remain audible. We varied in real-time the fraction of original audio mixed from 0% to 40% (0% if all of the input is estimated as speech, up to 40% as more of the input is estimated as non-speech) based on the estimate from the open-source YAMNet classifier on every ~1 second window of audio of whether the input is speech or non-speech.

The Conv-TasNet Speech Enhancement Model
To implement a speech enhancement module that suppresses non-speech sounds, such as noise and music, we use the Conv-TasNet model, which can separate different kinds of sounds. To start, the raw audio waveforms are transformed and processed into a form that can be used by a neural network. The model transforms short, 2.5 millisecond frames of input audio with a learnable analysis transform to generate features optimized for sound separation. The network then produces two “masks” from those features: one mask for speech and one mask for noise. These masks indicate the degree to which each feature corresponds to either speech or noise. Separated speech and noise are reconstructed back to the audio domain by multiplying the masks with the analysis features, applying a synthesis transform back to audio-domain frames, and stitching the resulting short frames together. As a final step, the speech and noise estimates are processed by a mixture consistency layer, which improves the quality of the estimated waveforms by ensuring that they sum up to the original input mixture waveform.

Block diagram of the speech enhancement system, which is based on Conv-TasNet.

The model is both causal and low latency: for each 2.5 milliseconds of input audio, the model produces estimates of separated speech and noise, and thus could be used in real-time. For the hackathon, to demonstrate what could be possible with increased compute power in future hardware, we chose to use a model variant with 2.9 million parameters. This model size is too large to be practically implemented in a CI today, but demonstrates what kind of performance would be possible with more capable hardware in the future.

Listening to the Results
As we optimized our models and overall solution, we used the hackathon-provided vocoder (which required a fixed temporal spacing of electrical pulses) to produce audio simulating what CI users might perceive. We then conducted blind A-B listening tests as typical hearing users.

Listening to the vocoder simulations below, the speech in the reconstructed sounds — from the vocoder processing the electrodograms — is reasonably intelligible when the input sound doesn’t contain too much background noise, however there is still room to improve the clarity of the speech. Our submission performed well in the speech-in-noise category and achieved second place overall.

A bottleneck on quality is that the fixed temporal spacing of stimulation pulses sacrifices fine-time structure in the audio. A change to the processing to produce pulses timed to peaks in the filtered sound waveforms captures more information about the pitch and structure of sound than is conventionally represented in implant stimulation patterns.

It’s important to note that this second vocoder output is overly optimistic about how well it might sound to a real CI user. For instance, the simple vocoder used here does not model how current spread in the cochlea blurs the stimulus, making it harder to resolve different frequencies. But this at least suggests that preserving fine-time structure is valuable and that the electrodogram itself is not the bottleneck.

Ideally, all processing approaches would be evaluated by a broad range of CI users, with the electrodograms implemented directly on their CIs rather than relying upon vocoder simulations.

Conclusion and a Call to Collaborate
We are planning to follow up on this experience in two main directions. First, we plan to explore the application of noise suppression to other hearing-accessibility modalities, including hearing aids, transcription, and vibrotactile sensory substitution. Second, we’ll take a deeper dive into the creation of electrodogram patterns for cochlear implants, exploiting fine temporal structure that is not accommodated in the usual CIS (continous interleaved sampling) patterns that are standard in the industry. According to Louizou: “It remains a puzzle how some single-channel patients can perform so well given the limited spectral information they receive”. Therefore, using fine temporal structure might be a critical step towards achieving an improved CI experience.

Google is committed to building technology with and for people with disabilities. If you are interested in collaborating to improve the state of the art in cochlear implants (or hearing aids), please reach out to ci-collaborators@googlegroups.com.

Acknowledgements
We would like to thank the Cochlear Impact hackathon organizers for giving us this opportunity and partnering with us. The participating team within Google is Samuel J. Yang, Scott Wisdom, Pascal Getreuer, Chet Gnegy, Mihajlo Velimirović, Sagar Savla, and Richard F. Lyon with guidance from Dan Ellis and Manoj Plakal.

Posted by Subhrajit Roy, Research Scientist and Diana Mincu, Research Software Engineer, Google Research

The intensive care unit (ICU) of a hospital looks after the most medically vulnerable patients, many of whom require organ support, such as mechanical ventilation or dialysis. While always critical, the demand on ICU services during the COVID-19 pandemic has further underscored the importance of data-driven decision-making in healthcare. Furthermore, the ability to accurately predict the clinical outcomes of ICU patients has the potential to guide therapy and may inform decisions about most effective care, including staffing and triage support.

Applying machine learning (ML) to electronic health records (EHRs) has shown promise in predicting clinical outcomes. However, many of these ML models are based on single-task learning (ST), where the models are trained only to predict a specific adverse event, such as an organ dysfunction or the need for a life-support intervention. Of greater benefit would be to train multi-task models, which take into account a variety of competing risks along with the interdependencies between organ systems that factor into patient outcomes in a realistic setting.

In “Multi-task prediction of organ dysfunction in the ICU using sequential sub-network routing”, we propose a multi-task learning (MTL) architecture, called Sequential Sub-Network Routing (SeqSNR), that better captures the complexity of a realistic setting. Inspired by a clinician’s holistic approach to diagnosing problems, SeqSNR is designed to use flexible parameter sharing and routing to find related tasks and encourage cross-learning between them. We successfully applied SeqSNR to the task of continuous adverse event prediction in an ICU setting and showed advantages over single-task and naïve multi-tasking, especially in low training data scenarios.

Data and Labels
In this study, we used the freely available, open access, de-identified MIMIC-III EHR dataset, which includes a patient cohort consisting of 36,498 adults across 52,038 critical care admissions at the Beth Israel Deaconess Medical Center between 2001 and 2012. Similar to our previous studies, we employed a version of the MIMIC-III dataset that was mapped to the Fast Healthcare Interoperability Resource (FHIR) standard and used a comprehensive set of features, including a sequence of vital signs, laboratory results, past medications, procedures, diagnoses, and more.

The MIMIC-III database contains multi-modal recordings from ICU patients. Unlike most datasets in ML, the input and targets are often not explicitly defined and must be inferred from the data. So, using a combination of automated rule-based methods and clinical review, we defined a suite of diverse endpoints, including critical care interventions, specific organ dysfunctions, and overall patient outcomes.

The task given to the model was to predict the onset of a selection of adverse events within 24–48 hours for every hour after a patient’s admission into the ICU. The defined adverse events included acute kidney injury (AKI), continuous renal replacement therapy (CRRT) dialysis, administration of vasopressors and inotropes, mechanical ventilation (MV), mortality, and remaining length of stay (LoS).

The SeqSNR Algorithm
While multi-task learning captures the interdependencies between organ systems and balances competing risks, it can be challenging to implement successfully. In practice, jointly-trained tasks often impair one another, an effect called “negative transfer”. The intuition behind SeqSNR was that modular ‘sub-networks’ would mitigate this issue by automatically optimizing how information is shared across multiple tasks.

SeqSNR is a time series adaptation of the SNR architecture and is a combination of a deep embedding layer followed by stacked recurrent neural network (RNN) layers. Modularisation is achieved by splitting both the embedding layer and the RNN stack into multiple modules connected by routing variables that are learned during the training phase. The routing connections are always created between blocks in one layer and the next. This approach minimizes negative transfer by ensuring that data of low relevance to a particular task layer is filtered out. In essence, this means that each task utilizes a different path through the model.

A high-level overview of the SeqSNR architecture.

Findings
SeqSNR shows a modest improvement in discriminative performance overall relative to single-task and naïve multitasking. However, it’s performance improvement is more significant in scenarios with few training labels.

Because the prevalence of different outcomes varied widely in the dataset (e.g. ~38% of patients had MV, but CRRT dialysis is present for only ~3%), many accuracy metrics are not suitable. Instead, we report the area under the precision recall curve (AU PRC), which is more reliable given imbalanced data. Moreover, we performed the Wilcoxon Signed Rank Tests to draw statistically significant conclusions for pairwise comparisons of ST learning, shared-bottom (SB) multi-task learning (i.e., naïve multi-task learning), and SeqSNR across bootstrapped samples from the held-out test set. The performance differences between the three architectures were modest, but SeqSNR outperformed both ST and SB in four out of six tasks (p-values are reported in the paper).

Comparison of single task (ST), shared bottom (SB) and SeqSNR performance on the MIMIC-III dataset.

Label Efficiency
We hypothesized that multi-task learning could assist in low-data scenarios by using easy-to-label auxiliary tasks to boost the performance of the main tasks. We formulated prediction tasks with only a portion of the training labels available for the primary prediction task, but kept the entire dataset for the “helper tasks”. The latter are chosen because they are reliably encoded in the EHR and are straightforward to timestamp. An example of such a helper task is length of stay, since the start and end of admissions are accurately timestamped in MIMIC-III. On the other hand, the start and end of mechanical ventilation events are not reliably timestamped. So, we defined a set of rules based on expert-defined heuristics to determine the ventilation times using multiple sources of mechanical ventilator–related settings along with physiological measurements in the EHR dataset that are indicative of MV.

The development of these rules for a new clinical endpoint was time-consuming and involved manual review of the dataset by experts. The difficulty in exhaustively labeling the dataset led us to test the model performance with only 1–10% of the data labeled, which resulted in a decline in model performance. The “helper tasks” are useful in this scenario since they are 100% labeled and can be used with the primary tasks (1–10% labeled) to jointly train the multi-task model for improved overall performance.

We chose AKI, mechanical ventilation, CRRT Dialysis, and vasoactive medications as primary endpoints using 1%, 5%, and 10% of the training labels, along with 100% of labels for the helper tasks — labs and vitals, mortality, and LoS. Performance of both ST and SeqSNR decreased as the percentage of labels for the primary endpoint was reduced, but SeqSNR outperformed ST across all tasks and all training data reduction percentages, with a statistically significant boost in performance for all cases.

Label efficiency results showing the discriminative performance when the training dataset for the primary endpoint is reduced to 1%, 5% and 10% while the helper tasks have access to all training labels.

This is a useful finding, given the difficulties of annotating endpoint labels in EHR datasets, which frequently necessitates human evaluation by doctors. The ability to use numerous endpoints, some of which may be easier to label (like duration of stay or mortality), could lessen the need for manual curation on more difficult endpoints that are annotated differently (like mechanical ventilation).

Subgroup Performance
While the version of the MIMIC-III dataset used contained labels for gender and age, it did not contain information on race and the information on ethnicity was limited. We computed the performance of all selected models across age and gender subgroups. We observed that in the scenarios with few instances in the dataset, the MTL models (both SB models and SeqSNR) often outperform ST. Even though there are exceptions, on average all models seem to be relatively balanced across age and gender subgroups. We invite the reader to refer to the supplemental section of our paper for a detailed performance breakdown.

Next Steps
This work is a proof of concept for SeqSNR on a set of canonical EHR prediction tasks. The code for this architecture is publicly available here. And will hopefully stimulate further research in EHR multi-tasking and other deep learning architectures inspired by clinical reasoning.

In future, it will be important to evaluate the performance of SeqSNR on different combinations of tasks, different time horizons and different datasets. One other area of potential growth in this project is to expand subgroup analysis by including datasets with additional population information, race, ethnicity, etc. Another area we are exploring is expanding subgroup analysis by including datasets with additional population information, such as race, ethnicity, etc. We also emphasize that these are prototype models designed to showcase methodologies, and more rigorous evaluation would be needed to bring these tools into deployment.

Acknowledgements
This work involved collaborative efforts from a multidisciplinary team of researchers, software engineers, clinicians, and cross-functional contributors. We thank our co-authors: Eric Loreaux, Anne Mottram, Ivan Protsyuk, Natalie Harris, Sebastien Baur, Yuan Xue, Jessica Schrouff, Ali Connell, Alan Karthikesalingam, Martin Seneviratne from Google, Nenad Tomasev from Deepmind, and Hugh Montgomery from University College London. We also thank Zhe Zhao from Google Research and Kathryn Rough, Cian Hughes, Megumi Morigami and Doris Wong from Google Health for their input and review, and the MIMIC team for curating this open access dataset for the research community.