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Advancing NLP with Efficient Projection-Based Model Architectures

Deep neural networks have radically transformed natural language processing (NLP) in the last decade, primarily through their application in data centers using specialized hardware. However, issues such as preserving user privacy, eliminating network latency, enabling offline functionality, and reducing operation costs have rapidly spurred the development of NLP models that can be run on-device rather than in data centers. Yet mobile devices have limited memory and processing power, which requires models running on them to be small and efficient — without compromising quality.

Last year, we published a neural architecture called PRADO, which at the time achieved state-of-the-art performance on many text classification problems, using a model with less than 200K parameters. While most models use a fixed number of parameters per token, the PRADO model used a network structure that required extremely few parameters to learn the most relevant or useful tokens for the task.

Today we describe a new extension to the model, called pQRNN, which advances the state of the art for NLP performance with a minimal model size. The novelty of pQRNN is in how it combines a simple projection operation with a quasi-RNN encoder for fast, parallel processing. We show that the pQRNN model is able to achieve BERT-level performance on a text classification task with orders of magnitude fewer number of parameters.

What Makes PRADO Work?
When developed a year ago, PRADO exploited NLP domain-specific knowledge on text segmentation to reduce the model size and improve the performance. Normally, the text input to NLP models is first processed into a form that is suitable for the neural network, by segmenting text into pieces (tokens) that correspond to values in a predefined universal dictionary (a list of all possible tokens). The neural network then uniquely identifies each segment using a trainable parameter vector, which comprises the embedding table. However, the way in which text is segmented has a significant impact on the model performance, size, and latency. The figure below shows the spectrum of approaches used by the NLP community and their pros and cons.

Since the number of text segments is such an important parameter for model performance and compression, it raises the question of whether or not an NLP model needs to be able to distinctly identify every possible text segment. To answer this question we look at the inherent complexity of NLP tasks.

Only a few NLP tasks (e.g., language models and machine translation) need to know subtle differences between text segments and thus need to be capable of uniquely identifying all possible text segments. In contrast, the majority of other tasks can be solved by knowing a small subset of these segments. Furthermore, this subset of task-relevant segments will likely not be the most frequent, as a significant fraction of segments will undoubtedly be dedicated to articles, such as a, an, the, etc., which for many tasks are not necessarily critical. Hence, allowing the network to determine the most relevant segments for a given task results in better performance. In addition, the network does not need to be able to uniquely identify these segments, but only needs to recognize clusters of text segments. For example, a sentiment classifier just needs to know segment clusters that are strongly correlated to the sentiment in the text.

Leveraging these insights, PRADO was designed to learn clusters of text segments from words rather than word pieces or characters, which enabled it to achieve good performance on low-complexity NLP tasks. Since word units are more meaningful, and yet the most relevant words for most tasks are reasonably small, many fewer model parameters are needed to learn such a reduced subset of relevant word clusters.

Improving PRADO
Building on the success of PRADO, we developed an improved NLP model, called pQRNN. This model is composed of three building blocks, a projection operator that converts tokens in text to a sequence of ternary vectors, a dense bottleneck layer and a stack of QRNN encoders.

The implementation of the projection layer in pQRNN is identical to that used in PRADO and helps the model learn the most relevant tokens without a fixed set of parameters to define them. It first fingerprints the tokens in the text and converts it to a ternary feature vector using a simple mapping function. This results in a ternary vector sequence with a balanced symmetric distribution that uniquely represents the text. This representation is not directly useful since it does not have any information needed to solve the task of interest and the network has no control over this representation. We combine it with a dense bottleneck layer to allow the network to learn a per word representation that is relevant for the task at hand. The representation resulting from the bottleneck layer still does not take the context of the word into account. We learn a contextual representation by using a stack of bidirectional QRNN encoders. The result is a network that is capable of learning a contextual representation from just text input without employing any kind of preprocessing.

Performance
We evaluated pQRNN on the civil_comments dataset and compared it with the BERT model on the same task. Simply because the model size is proportional to the number of parameters, pQRNN is much smaller than BERT. But in addition, pQRNN is quantized, further reducing the model size by a factor of 4x. The public pretrained version of BERT performed poorly on the task hence the comparison is done to a BERT version that is pretrained on several different relevant multilingual data sources to achieve the best possible performance.

We capture the area under the curve (AUC) for the two models. Without any kind of pre-training and just trained on the supervised data, the AUC for pQRNN is 0.963 using 1.3 million quantized (8-bit) parameters. With pre-training on several different data sources and fine-tuning on the supervised data, the BERT model gets 0.976 AUC using 110 million floating point parameters.

Conclusion
Using our previous generation model PRADO, we have demonstrated how it can be used as the foundation for the next generation of state-of-the-art light-weight text classification models. We present one such model, pQRNN, and show that this new architecture can nearly achieve BERT-level performance, despite using 300x fewer parameters and being trained on only the supervised data. To stimulate further research in this area, we have open-sourced the PRADO model and encourage the community to use it as a jumping off point for new model architectures.

Acknowledgements
We thank Yicheng Fan, Márius Šajgalík, Peter Young and Arun Kandoor for contributing to the open sourcing effort and helping improve the models. We would also like to thank Amarnag Subramanya, Ashwini Venkatesh, Benoit Jacob, Catherine Wah, Dana Movshovitz-Attias, Dang Hien, Dmitry Kalenichenko, Edgar Gonzàlez i Pellicer, Edward Li, Erik Vee, Evgeny Livshits, Gaurav Nemade, Jeffrey Soren, Jeongwoo Ko, Julia Proskurnia, Rushin Shah, Shirin Badiezadegan, Sidharth KV, Victor Cărbune and the Learn2Compress team for their support. We would like to thank Andrew Tomkins and Patrick Mcgregor for sponsoring this research project.

Source: Google AI Blog


Advancing NLP with Efficient Projection-Based Model Architectures

Deep neural networks have radically transformed natural language processing (NLP) in the last decade, primarily through their application in data centers using specialized hardware. However, issues such as preserving user privacy, eliminating network latency, enabling offline functionality, and reducing operation costs have rapidly spurred the development of NLP models that can be run on-device rather than in data centers. Yet mobile devices have limited memory and processing power, which requires models running on them to be small and efficient — without compromising quality.

Last year, we published a neural architecture called PRADO, which at the time achieved state-of-the-art performance on many text classification problems, using a model with less than 200K parameters. While most models use a fixed number of parameters per token, the PRADO model used a network structure that required extremely few parameters to learn the most relevant or useful tokens for the task.

Today we describe a new extension to the model, called pQRNN, which advances the state of the art for NLP performance with a minimal model size. The novelty of pQRNN is in how it combines a simple projection operation with a quasi-RNN encoder for fast, parallel processing. We show that the pQRNN model is able to achieve BERT-level performance on a text classification task with orders of magnitude fewer number of parameters.

What Makes PRADO Work?
When developed a year ago, PRADO exploited NLP domain-specific knowledge on text segmentation to reduce the model size and improve the performance. Normally, the text input to NLP models is first processed into a form that is suitable for the neural network, by segmenting text into pieces (tokens) that correspond to values in a predefined universal dictionary (a list of all possible tokens). The neural network then uniquely identifies each segment using a trainable parameter vector, which comprises the embedding table. However, the way in which text is segmented has a significant impact on the model performance, size, and latency. The figure below shows the spectrum of approaches used by the NLP community and their pros and cons.

Since the number of text segments is such an important parameter for model performance and compression, it raises the question of whether or not an NLP model needs to be able to distinctly identify every possible text segment. To answer this question we look at the inherent complexity of NLP tasks.

Only a few NLP tasks (e.g., language models and machine translation) need to know subtle differences between text segments and thus need to be capable of uniquely identifying all possible text segments. In contrast, the majority of other tasks can be solved by knowing a small subset of these segments. Furthermore, this subset of task-relevant segments will likely not be the most frequent, as a significant fraction of segments will undoubtedly be dedicated to articles, such as a, an, the, etc., which for many tasks are not necessarily critical. Hence, allowing the network to determine the most relevant segments for a given task results in better performance. In addition, the network does not need to be able to uniquely identify these segments, but only needs to recognize clusters of text segments. For example, a sentiment classifier just needs to know segment clusters that are strongly correlated to the sentiment in the text.

Leveraging these insights, PRADO was designed to learn clusters of text segments from words rather than word pieces or characters, which enabled it to achieve good performance on low-complexity NLP tasks. Since word units are more meaningful, and yet the most relevant words for most tasks are reasonably small, many fewer model parameters are needed to learn such a reduced subset of relevant word clusters.

Improving PRADO
Building on the success of PRADO, we developed an improved NLP model, called pQRNN. This model is composed of three building blocks, a projection operator that converts tokens in text to a sequence of ternary vectors, a dense bottleneck layer and a stack of QRNN encoders.

The implementation of the projection layer in pQRNN is identical to that used in PRADO and helps the model learn the most relevant tokens without a fixed set of parameters to define them. It first fingerprints the tokens in the text and converts it to a ternary feature vector using a simple mapping function. This results in a ternary vector sequence with a balanced symmetric distribution that uniquely represents the text. This representation is not directly useful since it does not have any information needed to solve the task of interest and the network has no control over this representation. We combine it with a dense bottleneck layer to allow the network to learn a per word representation that is relevant for the task at hand. The representation resulting from the bottleneck layer still does not take the context of the word into account. We learn a contextual representation by using a stack of bidirectional QRNN encoders. The result is a network that is capable of learning a contextual representation from just text input without employing any kind of preprocessing.

Performance
We evaluated pQRNN on the civil_comments dataset and compared it with the BERT model on the same task. Simply because the model size is proportional to the number of parameters, pQRNN is much smaller than BERT. But in addition, pQRNN is quantized, further reducing the model size by a factor of 4x. The public pretrained version of BERT performed poorly on the task hence the comparison is done to a BERT version that is pretrained on several different relevant multilingual data sources to achieve the best possible performance.

We capture the area under the curve (AUC) for the two models. Without any kind of pre-training and just trained on the supervised data, the AUC for pQRNN is 0.963 using 1.3 million quantized (8-bit) parameters. With pre-training on several different data sources and fine-tuning on the supervised data, the BERT model gets 0.976 AUC using 110 million floating point parameters.

Conclusion
Using our previous generation model PRADO, we have demonstrated how it can be used as the foundation for the next generation of state-of-the-art light-weight text classification models. We present one such model, pQRNN, and show that this new architecture can nearly achieve BERT-level performance, despite using 300x fewer parameters and being trained on only the supervised data. To stimulate further research in this area, we have open-sourced the PRADO model and encourage the community to use it as a jumping off point for new model architectures.

Acknowledgements
We thank Yicheng Fan, Márius Šajgalík, Peter Young and Arun Kandoor for contributing to the open sourcing effort and helping improve the models. We would also like to thank Amarnag Subramanya, Ashwini Venkatesh, Benoit Jacob, Catherine Wah, Dana Movshovitz-Attias, Dang Hien, Dmitry Kalenichenko, Edgar Gonzàlez i Pellicer, Edward Li, Erik Vee, Evgeny Livshits, Gaurav Nemade, Jeffrey Soren, Jeongwoo Ko, Julia Proskurnia, Rushin Shah, Shirin Badiezadegan, Sidharth KV, Victor Cărbune and the Learn2Compress team for their support. We would like to thank Andrew Tomkins and Patrick Mcgregor for sponsoring this research project.

Source: Google AI Blog


Advancing NLP with Efficient Projection-Based Model Architectures

Deep neural networks have radically transformed natural language processing (NLP) in the last decade, primarily through their application in data centers using specialized hardware. However, issues such as preserving user privacy, eliminating network latency, enabling offline functionality, and reducing operation costs have rapidly spurred the development of NLP models that can be run on-device rather than in data centers. Yet mobile devices have limited memory and processing power, which requires models running on them to be small and efficient — without compromising quality.

Last year, we published a neural architecture called PRADO, which at the time achieved state-of-the-art performance on many text classification problems, using a model with less than 200K parameters. While most models use a fixed number of parameters per token, the PRADO model used a network structure that required extremely few parameters to learn the most relevant or useful tokens for the task.

Today we describe a new extension to the model, called pQRNN, which advances the state of the art for NLP performance with a minimal model size. The novelty of pQRNN is in how it combines a simple projection operation with a quasi-RNN encoder for fast, parallel processing. We show that the pQRNN model is able to achieve BERT-level performance on a text classification task with orders of magnitude fewer number of parameters.

What Makes PRADO Work?
When developed a year ago, PRADO exploited NLP domain-specific knowledge on text segmentation to reduce the model size and improve the performance. Normally, the text input to NLP models is first processed into a form that is suitable for the neural network, by segmenting text into pieces (tokens) that correspond to values in a predefined universal dictionary (a list of all possible tokens). The neural network then uniquely identifies each segment using a trainable parameter vector, which comprises the embedding table. However, the way in which text is segmented has a significant impact on the model performance, size, and latency. The figure below shows the spectrum of approaches used by the NLP community and their pros and cons.

Since the number of text segments is such an important parameter for model performance and compression, it raises the question of whether or not an NLP model needs to be able to distinctly identify every possible text segment. To answer this question we look at the inherent complexity of NLP tasks.

Only a few NLP tasks (e.g., language models and machine translation) need to know subtle differences between text segments and thus need to be capable of uniquely identifying all possible text segments. In contrast, the majority of other tasks can be solved by knowing a small subset of these segments. Furthermore, this subset of task-relevant segments will likely not be the most frequent, as a significant fraction of segments will undoubtedly be dedicated to articles, such as a, an, the, etc., which for many tasks are not necessarily critical. Hence, allowing the network to determine the most relevant segments for a given task results in better performance. In addition, the network does not need to be able to uniquely identify these segments, but only needs to recognize clusters of text segments. For example, a sentiment classifier just needs to know segment clusters that are strongly correlated to the sentiment in the text.

Leveraging these insights, PRADO was designed to learn clusters of text segments from words rather than word pieces or characters, which enabled it to achieve good performance on low-complexity NLP tasks. Since word units are more meaningful, and yet the most relevant words for most tasks are reasonably small, many fewer model parameters are needed to learn such a reduced subset of relevant word clusters.

Improving PRADO
Building on the success of PRADO, we developed an improved NLP model, called pQRNN. This model is composed of three building blocks, a projection operator that converts tokens in text to a sequence of ternary vectors, a dense bottleneck layer and a stack of QRNN encoders.

The implementation of the projection layer in pQRNN is identical to that used in PRADO and helps the model learn the most relevant tokens without a fixed set of parameters to define them. It first fingerprints the tokens in the text and converts it to a ternary feature vector using a simple mapping function. This results in a ternary vector sequence with a balanced symmetric distribution that uniquely represents the text. This representation is not directly useful since it does not have any information needed to solve the task of interest and the network has no control over this representation. We combine it with a dense bottleneck layer to allow the network to learn a per word representation that is relevant for the task at hand. The representation resulting from the bottleneck layer still does not take the context of the word into account. We learn a contextual representation by using a stack of bidirectional QRNN encoders. The result is a network that is capable of learning a contextual representation from just text input without employing any kind of preprocessing.

Performance
We evaluated pQRNN on the civil_comments dataset and compared it with the BERT model on the same task. Simply because the model size is proportional to the number of parameters, pQRNN is much smaller than BERT. But in addition, pQRNN is quantized, further reducing the model size by a factor of 4x. The public pretrained version of BERT performed poorly on the task hence the comparison is done to a BERT version that is pretrained on several different relevant multilingual data sources to achieve the best possible performance.

We capture the area under the curve (AUC) for the two models. Without any kind of pre-training and just trained on the supervised data, the AUC for pQRNN is 0.963 using 1.3 million quantized (8-bit) parameters. With pre-training on several different data sources and fine-tuning on the supervised data, the BERT model gets 0.976 AUC using 110 million floating point parameters.

Conclusion
Using our previous generation model PRADO, we have demonstrated how it can be used as the foundation for the next generation of state-of-the-art light-weight text classification models. We present one such model, pQRNN, and show that this new architecture can nearly achieve BERT-level performance, despite using 300x fewer parameters and being trained on only the supervised data. To stimulate further research in this area, we have open-sourced the PRADO model and encourage the community to use it as a jumping off point for new model architectures.

Acknowledgements
We thank Yicheng Fan, Márius Šajgalík, Peter Young and Arun Kandoor for contributing to the open sourcing effort and helping improve the models. We would also like to thank Amarnag Subramanya, Ashwini Venkatesh, Benoit Jacob, Catherine Wah, Dana Movshovitz-Attias, Dang Hien, Dmitry Kalenichenko, Edgar Gonzàlez i Pellicer, Edward Li, Erik Vee, Evgeny Livshits, Gaurav Nemade, Jeffrey Soren, Jeongwoo Ko, Julia Proskurnia, Rushin Shah, Shirin Badiezadegan, Sidharth KV, Victor Cărbune and the Learn2Compress team for their support. We would like to thank Andrew Tomkins and Patrick Mcgregor for sponsoring this research project.

Source: Google AI Blog


Improving the Accuracy of Genomic Analysis with DeepVariant 1.0

Sequencing genomes involves sampling short pieces of the DNA from the ~6 billion pairs of nucleobases — i.e., adenine (A), thymine (T), guanine (G), and cytosine (C) — we inherit from our parents. Genome sequencing is enabled by two key technologies: DNA sequencers (hardware) that "read" relatively small fragments of DNA, and variant callers (software) that combine the reads to identify where and how an individual's genome differs from a reference genome, like the one assembled in the Human Genome Project. Such variants may be indicators of genetic disorders, such as an elevated risk for breast cancer, pulmonary arterial hypertension, or neurodevelopmental disorders.

In 2017, we released DeepVariant, an open-source tool which identifies genome variants in sequencing data using a convolutional neural network (CNN). The sequencing process begins with a physical sample being sequenced by any of a handful of instruments, depending on the end goal of the sequencing. The raw data, which consists of numerous reads of overlapping fragments of the genome, are then mapped to a reference genome. DeepVariant analyzes these mappings to identify variant locations and distinguish them from sequencing errors.

Soon after it was first published in 2018, DeepVariant underwent a number of updates and improvements, including significant changes to improve accuracy for whole exome sequencing and polymerase chain reaction (PCR) sequencing.

We are now releasing DeepVariant v1.0, which incorporates a large number of improvements for all sequencing types. DeepVariant v1.0 is an improved version of our submission to the PrecisionFDA v2 Truth Challenge, which achieved Best Overall accuracy for 3 of 4 instrument categories. Compared to previous state-of-the-art models, DeepVariant v1.0 significantly reduces the errors for widely-used sequencing data types, including Illumina and Pacific Biosciences. In addition, through a collaboration with the UCSC Genomics Institute, we have also released a model that combines DeepVariant with the UCSC’s PEPPER method, called PEPPER-DeepVariant, which extends coverage to Oxford Nanopore data for the first time.

Sequencing Technologies and DeepVariant
For the last decade, the majority of sequence data were generated using Illumina instruments, which produce short (75-250 bases) and accurate sequences. In recent years, new technologies have become available that can sequence much longer pieces, including Pacific Biosciences, which can produce long and accurate sequences up to ~15,000 bases in length, and Oxford Nanopore, which can produce reads up to 1 million bases long, but with higher error rates. The particular type of sequencing data a researcher might use depends on the ultimate use-case.

Because DeepVariant is a deep learning method, we can quickly re-train it for these new instrument types, ensuring highly accurate sequence identification. Accuracy is important because a missed variant call could mean missing the causal variant for a disorder, while a false positive variant call could lead to identifying an incorrect one. Earlier state-of-the-art methods could reach ~99.1% accuracy (~73,000 errors) on a 35-fold coverage Illumina whole genome, whereas an early version of DeepVariant (v0.10) had ~99.4% accuracy (46,000 errors), corresponding to a 38% error reduction. DeepVariant v1.0 reduces Illumina errors by another ~22% and PacBio errors by another ~52% relative to the last DeepVariant release (v0.10).

DeepVariant Overview
DeepVariant is a convolutional neural network (CNN) that treats the task of identifying genetic variants as an image classification problem. DeepVariant constructs tensors, essentially multi-channel images, where each channel represents an aspect of the sequence, such as the bases in the sequence (called read base), the quality of alignment between different reads (mapping quality), whether a given read supports an alternate allele (read supports variant), etc. It then analyzes these data and outputs three genotype likelihoods, corresponding to how many copies (0, 1, or 2) of a given alternate allele are present.

Example of DeepVariant data. Each row of pixels in each panel corresponds to a single read, i.e., a short genetic sequence. The top, middle, and bottom rows of panels present examples with a different number of variant alleles. Only two of the six data channels are shown: Read base — the pixel value is mapped to each of the four bases, A, C, G, or T; Read supports variant — white means that the read is consistent with a given allele and grey means it is not. Top: Classified by DeepVariant as a "2", which means that both chromosomes match the variant allele. Middle: Classified as a “1”, meaning that one chromosome matches the variant allele. Bottom: Classified as a “0”, implying that the variant allele is missing from both chromosomes.

Technical Improvements in DeepVariant v1.0
Because DeepVariant uses the same codebase for each data type, improvements apply to each of Illumina, PacBio, and Oxford Nanopore. Below, we show the numbers for Illumina and PacBio for two types of small variants: SNPs (single nucleotide polymorphisms, which change a single base without changing sequence length) and INDELs (insertions and deletions).

  • Training on an extended truth set

    The Genome in a Bottle consortium from the National Institute of Standards and Technology (NIST) creates gold-standard samples with known variants covering the regions of the genome. These are used as labels to train DeepVariant. Using long-read technologies the Genome in a Bottle expanded the set of confident variants, increasing the regions described by the standard set from 85% of the genome to 92% of it. These more difficult regions were already used in training the PacBio models, and including them in the Illumina models reduced errors by 11%. By relaxing the filter for reads of lower mapping quality, we further reduced errors by 4% for Illumina and 13% for PacBio.

  • Haplotype sorting of long reads

    We inherit one copy of DNA from our mother and another from our father. PacBio and Oxford Nanopore sequences are long enough to separate sequences by parental origin, which is called a haplotype. By providing this information to the neural network, DeepVariant improves its identification of random sequence errors and can better determine whether a variant has a copy from one or both parents.

  • Re-aligning reads to the alternate (ALT) allele

    DeepVariant uses input sequence fragments that have been aligned to a reference genome. The optimal alignment for variants that include insertions or deletions could be different if the aligner knew they were present. To capture this information, we implemented an additional alignment step relative to the candidate variant. The figure below shows an additional second row where the reads are aligned to the candidate variant, which is a large insertion. You can see sequences that abruptly stop in the first row can now be fully aligned, providing additional information.

    Example of DeepVariant data with realignment to ALT allele. DeepVariant is presented the information in both rows of data for the same example. Only two of the six data channels are shown: Read base (channel #1) and Read supports variant (channel #5). Top: Shows the reads aligned to the reference (in DeepVariant v0.10 and earlier this is all DeepVariant sees). Bottom: Shows the reads aligned to the candidate variant, in this case a long insertion of sequence). The red arrow indicates where the inserted sequence begins.
  • Use a small network to post-process outputs

    Variants can have multiple alleles, with a different base inherited from each parent. DeepVariant’s classifier only generates a probability for one potential variant at a time. In previous versions, simple hand-written rules converted the probabilities into a composite call, but these rules failed in some edge cases. In addition, it also separated the way a final call was made from the backpropagation to train the network. By adding a small, fully-connected neural network to the post-processing step, we are able to better handle these tricky multi-allelic cases.

  • Adding data to train the release model

    The timeframe for the competition was compressed, so we trained only with data similar to the challenge data (PCR-Free NovaSeq) to speed model training. In our production releases, we seek high accuracy for multiple instruments as well as PCR+ preparations. Training with data from these diverse classes helps the model generalize, so our DeepVariant v1.0 release model outperforms the one submitted.

The charts below show the error reduction achieved by each improvement.

Training a Hybrid model
DeepVariant v1.0 also includes a hybrid model for PacBio and Illumina reads. In this case, the model leverages the strengths of both input types, without needing new logic.

Example of DeepVariant merging data from both PacBio and Illumina. Only two of the six data channels are shown: Read base (channel #1) and Read supports variant (channel #5). The longer PacBio reads (at the upper part of the image) span the region being called entirely, while the shorter Illumin reads span only a portion of the region.

We observed no change in SNP errors, suggesting that PacBio reads are strictly superior for SNP calling. We observed a further 49% reduction in Indel errors relative to the PacBio model, suggesting that the Indel error modes of Illumina and PacBio HiFi can be used in a complementary manner.

PEPPER-Deepvariant: A Pipeline for Oxford Nanopore Data Using DeepVariant
Until the PrecisionFDA competition, a DeepVariant model was not available for Oxford Nanopore data, because the higher base error rate created too many candidates for DeepVariant to classify. We partnered with the UC Santa Cruz Genomics Institute, which has extensive expertise with Nanopore data. They had previously trained a deep learning method called PEPPER, which could narrow down the candidates to a more tractable number. The larger neural network of DeepVariant can then accurately characterize the remaining candidates with a reasonable runtime.

The combined PEPPER-DeepVariant pipeline with the Oxford Nanopore model is open-source and available on GitHub. This pipeline was able to achieve a superior SNP calling accuracy to DeepVariant Illumina on the PrecisionFDA challenge, which is the first time anyone has shown Nanopore outperforming Illumina in this way.

Conclusion
DeepVariant v1.0 isn’t the end of development. We look forward to working with the genomics community to further maximize the value of genomic data to patients and researchers.

Source: Google AI Blog


Improving Sparse Training with RigL

Modern deep neural network architectures are often highly redundant [1, 2, 3], making it possible to remove a significant fraction of connections without harming performance. The sparse neural networks that result have been shown to be more parameter and compute efficient compared to dense networks, and, in many cases, can significantly decrease wall clock inference times.

By far the most popular method for training sparse neural networks is pruning, (dense-to-sparse training) which usually requires first training a dense model, and then “sparsifying” it by cutting out the connections with negligible weights. However, this process has two limitations.

  1. The size of the largest trainable sparse model is limited by that of the largest trainable dense model. Even if sparse models are more parameter efficient, one cannot use pruning to train models that are larger and more accurate than the largest possible dense models.
  2. Pruning is inefficient, meaning that large amounts of computation must be performed for parameters that are zero valued or that will be zero during inference. Additionally, it remains unknown if the performance of the current best pruning algorithms are an upper bound on the quality of sparse models.
Training sparse networks from scratch, on the other hand, is efficient, however often achieves inferior performance compared to pruning.

In “Rigging the Lottery: Making All Tickets Winners”, presented at ICML 2020, we introduce RigL, an algorithm for training sparse neural networks that uses a fixed parameter count and computational cost throughout training, without sacrificing accuracy relative to existing dense-to-sparse training methods. The algorithm identifies which neurons should be active during training, which helps the optimization process to utilize the most relevant connections and results in better sparse solutions. An example of this is shown below, where, during the training of a multilayer perceptron (MLP) network on MNIST, our sparse network trained with RigL learns to focus on the center of the images, discarding the uninformative pixels from the edges. A Tensorflow implementation of our method along with three other baselines (SET, SNFS, SNIP) can be found at github.com/google-research/rigl.

Left: Average MNIST image. Right: Evolution of the connectivity of the input throughout the training of a 98% sparse, 2-layer MLP on MNIST. Training starts from a random sparse mask, where each input pixel has roughly six outgoing connections. Connections that originate from the edges do not exhibit meaningful gradients and are therefore replaced by more informative connections that originate from the center pixels.

RigL Overview
The RigL method starts with a network initialized with a random sparse topology. At regularly spaced intervals we remove a fraction of the connections with the smallest weight magnitudes. Such a strategy has been shown to have very little effect on the loss. RigL then activates new connections using instantaneous gradient information, i.e., without using past gradient information. After updating the connectivity, training continues with the updated network until the next scheduled update. Next, the system activates connections with large gradients, since these connections are expected to decrease the loss most quickly.

RigL begins with a random sparse initialization of the network. It then trains the network and trims out those connections with weak activations. Based on the gradients calculated for the new configuration, it grows new connections and trains again, repeating the cycle.

Evaluating Performance
By changing the connectivity of the neurons dynamically during training, RigL helps optimize to find better solutions. To demonstrate this, we restart training from a bad solution that exhibits poor accuracy and show that RigL's mask updates help the optimization achieve better loss compared to static training, in which connectivity of the sparse network remains the same.

Training loss of RigL and Static methods starting from the same static sparse solution, shown together with their final test accuracies.

The figure below summarizes the performance of various methods on training an 80% sparse ResNet-50 architecture. We compare RigL with two recent sparse training methods, SET and SNFS and three baseline training methods: Static, Small-Dense and Pruning. Two of these methods (SNFS and Pruning) require dense resources as they need to either train a large network or store the gradients of it. Overall, we observe that the performance of all methods improves with additional training time; thus, for each method we run extended training with up to 5x the training steps of the original 100 epochs.

As noted in a number of studies [4, 5, 6, 7], training a network with fixed sparsity from scratch (Static) leads to inferior performance compared to solutions found by pruning. Training a small, dense network (Small-Dense) with the same number of parameters gets better results than Static, but fails to match the performance of dynamic sparse models. Similarly, SET improves the performance over Small-Dense, but saturates at around 75% accuracy, revealing the limits of growing new connections randomly. Methods that use gradient information to grow new connections (RigL and SNFS) obtain higher accuracy in general, but RigL achieves the highest accuracy, while also consistently requiring fewer FLOPs (and memory footprint) than the other methods.

Performance of sparse training methods on training an 80% sparse ResNet-50 architecture with uniform sparsity distribution. Points at each curve correspond to the individual training runs with increasing training length. The number of FLOPs required to train a standard dense ResNet-50 along with its performance is indicated with a dashed red line. RigL matches the standard ResNet-50 performance, even though it is 5x smaller in size.

Observing the trend between extended training and performance, we compare the results using longer training runs. Within the interval considered (i.e., 1x-100x) RigL's performance constantly improves with additional training. RigL achieves state of art performance of 68.07% Top-1 accuracy at training with a 99% sparse ResNet-50 architecture. Similarly extended training of a 90% sparse MobileNet-v1 architecture with RigL achieves 70.55% Top-1 accuracy. Obtaining the same results with fewer training iterations is an exciting future research direction.

Effect of training time on RigL accuracy at training 99% sparse ResNet-50 (left) and 90% sparse MobileNets-v1 (right) architectures.

Other experiments include image classification on CIFAR-10 datasets and character-based language modelling using RNNs with the WikiText-103 dataset and can be found in the full paper.

Future Work
RigL is useful in three different scenarios:

  1. Improving the accuracy of sparse models intended for deployment.
  2. Improving the accuracy of large sparse models that can only be trained for a limited number of iterations.
  3. Combining with sparse primitives to enable training of extremely large sparse models which otherwise would not be possible.
The third scenario is unexplored due to the lack of hardware and software support for sparsity. Nonetheless, work continues [8, 9, 10] to improve the performance of sparse networks on current hardware and new types of hardware accelerators are expected to have better support for parameter sparsity [11, 12]. We hope RigL provides the tools to take advantage of, and motivation for, such advances.

AcknowledgementsWe would like to thank Eleni Triantafillou, Hugo Larochelle, Bart van Merrienboer, Fabian Pedregosa, Joan Puigcerver, Danny Tarlow, Nicolas Le Roux, Karen Simonyan for giving feedback on the preprint of the paper; Namhoon Lee for helping us verify and debug our SNIP implementation; Chris Jones for helping us discover and solve the distributed training bug; and Tom Small for creating the visualization of the algorithm.

Source: Google AI Blog


Imitation Learning in the Low-Data Regime

Reinforcement Learning (RL) is a paradigm for using trial-and-error to train agents to sequentially make decisions in complex environments, which has had great success in a number of domains, including games, robotics manipulation and chip design. Agents typically aim at maximizing the sum of the reward they collect in an environment, which can be based on a variety of parameters, including speed, curiosity, aesthetics and more. However, designing a specific RL reward function is a challenge since it can be hard to specify or too sparse. In such cases, imitation learning (IL) methods offer an alternative as they learn how to solve a task from expert demonstrations, rather than a carefully designed reward function. However, state-of-the-art IL methods rely on adversarial training, which uses min/max optimization procedures, making them algorithmically unstable and difficult to deploy.

In “Primal Wasserstein Imitation Learning” (PWIL), we introduce a new IL method, based on the primal form of the Wasserstein distance, also known as the earth mover’s distance, which does not rely on adversarial training. Using the MuJoCo suite of tasks, we demonstrate the efficacy of the PWIL method by imitating a simulated expert with a limited number of demonstrations (even a single example) and limited interactions with the environment.

Left: Demonstration of the algorithmic Humanoid “expert”, trained on the true reward of the task (which relates to speed). Right: Agent trained using PWIL on the expert demonstration.

Adversarial Imitation Learning
State-of-the-art adversarial IL methods operate similarly to generative adversarial networks (GANs) in which a generator (the policy) is trained to maximize the confusion of a discriminator (the reward) that itself is trained to differentiate between the agent’s state-action pairs and the expert’s. Adversarial IL methods boil down to a distribution matching problem, i.e., the problem of minimizing a distance between probability distributions in a metric space. However, just as GANs, adversarial IL methods rely on a min/max optimization problem and hence come with a number of training stability challenges.

Imitation Learning as Distribution Matching
The PWIL method is based on the formulation of IL as a distribution matching problem, in this case, the Wasserstein distance. The first step consists of inferring from the demonstrations a state-action distribution of the expert, the collection of relationships between the actions taken by the expert and the corresponding state of the environment. The goal is then to minimize the distance between the agent’s and the expert’s state-action distributions, through interactions with the environment. In contrast, PWIL is a non-adversarial method, enabling it to bypass the min/max optimization problem and directly minimize the Wasserstein distance between the agent’s and the expert’s state-action pair distributions.

Primal Wasserstein Imitation Learning
Computing the exact Wasserstein distance can be restrictive since one must wait until the end of a trajectory of the agent to calculate it, meaning that the rewards can be computed only when the agent is done interacting with the environment. To avoid this restriction, we use an upper bound on the distance instead, from which we can define a reward that we optimize using RL. We show that by doing so, we indeed recover expert behaviour and minimize the Wasserstein distance between the agent and the expert on a number of locomotion tasks of the MuJoCo simulator. While adversarial IL methods use a reward function from a neural network that must be optimized and re-estimated continuously as the agent interacts with the environment, PWIL defines a reward function offline from demonstrations, which does not change and is based on substantially fewer hyperparameters than adversarial IL approaches.

Training curves for PWIL on Humanoid. In green, the Wasserstein distance to the state-action distribution of the expert. In blue, the return (the sum of rewards collected) by the agent.

A Measure of Similarity for the True Imitation Learning Setting
As in numerous challenges in ML, a number of IL methods are evaluated on synthetic tasks, where one usually has access to the underlying reward function of the task and can measure similarity between the expert’s and the agent’s behaviour in terms of performance, which is the expected sum of rewards. A byproduct of PWIL is the creation of a metric that can compare expert behavior to an agent’s behavior for any IL method, without access to the true reward of the task. In this sense, we can use the Wasserstein distance in the true IL setting, not only on synthetic tasks.

Conclusion
In environments where interacting is costly (e.g., a real robot or a complex simulator), PWIL is a prime candidate not only because it can recover expert behaviour, but also because the reward function it defines is easy to tune and is defined without interactions with the environment. This opens multiple opportunities for future exploration, including deployment to real systems, extending PWIL to the setup where we have only access to demonstration states (rather than states and actions), and finally applying PWIL to visual based observations.

Acknowledgements
We thank our co-authors, Matthieu Geist and Olivier Pietquin; as well as Zafarali Ahmed, Adrien Ali Taïga, Gabriel Dulac-Arnold, Johan Ferret, Alexis Jacq and Saurabh Kumar for their feedback on the manuscript.

Source: Google AI Blog


The Technology Behind our Recent Improvements in Flood Forecasting

Flooding is the most common natural disaster on the planet, affecting the lives of hundreds of millions of people around the globe and causing around $10 billion in damages each year. Building on our work in previous years, earlier this week we announced some of our recent efforts to improve flood forecasting in India and Bangladesh, expanding coverage to more than 250 million people, and providing unprecedented lead time, accuracy and clarity.

To enable these breakthroughs, we have devised a new approach for inundation modeling, called a morphological inundation model, which combines physics-based modeling with machine learning (ML) to create more accurate and scalable inundation models in real-world settings. Additionally, our new alert-targeting model allows identifying areas at risk of flooding at unprecedented scale using end-to-end machine learning models and data that is publicly available globally. In this post, we also describe developments for the next generation of flood forecasting systems, called HydroNets (presented at ICLR AI for Earth Sciences and EGU this year), which is a new architecture specially built for hydrologic modeling across multiple basins, while still optimizing for accuracy at each location.

Forecasting Water Levels
The first step in a flood forecasting system is to identify whether a river is expected to flood. Hydrologic models (or gauge-to-gauge models) have long been used by governments and disaster management agencies to improve the accuracy and extend the lead time of their forecasts. These models receive inputs like precipitation or upstream gauge measurements of water level (i.e., the absolute elevation of the water above sea level) and output a forecast for the water level (or discharge) in the river at some time in the future.

The hydrologic model component of the flood forecasting system described in this week’s Keyword post doubled the lead time of flood alerts for areas covering more than 75 million people. These models not only increase lead time, but also provide unprecedented accuracy, achieving an R2 score of more than 99% across all basins we cover, and predicting the water level within a 15 cm error bound more than 90% of the time. Once a river is predicted to reach flood level, the next step in generating actionable warnings is to convert the river level forecast into a prediction for how the floodplain will be affected.

Morphological Inundation Modeling
In prior work, we developed high quality elevation maps based on satellite imagery, and ran physics-based models to simulate water flow across these digital terrains, which allowed warnings with unprecedented resolution and accuracy in data-scarce regions. In collaboration with our satellite partners, Airbus, Maxar and Planet, we have now expanded the elevation maps to cover hundreds of millions of square kilometers. However, in order to scale up the coverage to such a large area while still retaining high accuracy, we had to re-invent how we develop inundation models.

Inundation modeling estimates what areas will be flooded and how deep the water will be. This visualization conceptually shows how inundation could be simulated, how risk levels could be defined (represented by red and white colors), and how the model could be used to identify areas that should be warned (green dots).

Inundation modeling at scale suffers from three significant challenges. Due to the large areas involved and the resolution required for such models, they necessarily have high computational complexity. In addition, most global elevation maps don’t include riverbed bathymetry, which is important for accurate modeling. Finally, the errors in existing data, which may include gauge measurement errors, missing features in the elevation maps, and the like, need to be understood and corrected. Correcting such problems may require collecting additional high-quality data or fixing erroneous data manually, neither of which scale well.

Our new approach to inundation modeling, which we call a morphological model, addresses these issues by using several innovative tricks. Instead of modeling the complex behaviors of water flow in real time, we compute modifications to the morphology of the elevation map that allow one to simulate the inundation using simple physical principles, such as those describing hydrostatic systems.

First, we train a pure-ML model (devoid of physics-based information) to estimate the one-dimensional river profile from gauge measurements. The model takes as input the water level at a specific point on the river (the stream gauge) and outputs the river profile, which is the water level at all points in the river. We assume that if the gauge increases, the water level increases monotonically, i.e., the water level at other points in the river increases as well. We also assume that the absolute elevation of the river profile decreases downstream (i.e., the river flows downhill).

We then use this learned model and some heuristics to edit the elevation map to approximately “cancel out” the pressure gradient that would exist if that region were flooded. This new synthetic elevation map provides the foundation on which we model the flood behavior using a simple flood-fill algorithm. Finally, we match the resulting flooded map to the satellite-based flood extent with the original stream gauge measurement.

This approach abandons some of the realistic constraints of classical physics-based models, but in data scarce regions where existing methods currently struggle, its flexibility allows the model to automatically learn the correct bathymetry and fix various errors to which physics-based models are sensitive. This morphological model improves accuracy by 3%, which can significantly improve forecasts for large areas, while also allowing for much more rapid model development by reducing the need for manual modeling and correction.

Alert targeting
Many people reside in areas that are not covered by the morphological inundation models, yet access to accurate predictions are still urgently needed. To reach this population and to increase the impact of our flood forecasting models, we designed an end-to-end ML-based approach, using almost exclusively data that is globally publicly available, such as stream gauge measurements, public satellite imagery, and low resolution elevation maps. We train the model to use the data it is receiving to directly infer the inundation map in real time.

A direct ML approach from real-time measurements to inundation.

This approach works well “out of the box” when the model only needs to forecast an event that is within the range of events previously observed. Extrapolating to more extreme conditions is much more challenging. Nevertheless, proper use of existing elevation maps and real-time measurements can enable alerts that are more accurate than presently available for those in areas not covered by the more detailed morphological inundation models. Because this model is highly scalable, we were able to launch it across India after only a few months of work, and we hope to roll it out to many more countries soon.

Improving Water Levels Forecasting
In an effort to continue improving flood forecasting, we have developed HydroNets — a specialized deep neural network architecture built specifically for water levels forecasting — which allows us utilize some exciting recent advances in ML-based hydrology in a real-world operational setting. Two prominent features distinguish it from standard hydrologic models. First, it is able to differentiate between model components that generalize well between sites, such as the modeling of rainfall-runoff processes, and those that are specific to a given site, like the rating curve, which converts a predicted discharge volume into an expected water level. This enables the model to generalize well to different sites, while still fine-tuning its performance to each location. Second, HydroNets takes into account the structure of the river network being modeled, by training a large architecture that is actually a web of smaller neural networks, each representing a different location along the river. This allows neural networks that are modeling upstream sites to pass information encoded in embeddings to models of downstream sites, so that every model can know everything it needs without a drastic increase in parameters.

The animation below illustrates the structure and flow of information in HydroNets. The output from the modeling of upstream sub-basins is combined into a single representation of a given basin state. It is then processed by the shared model component, which is informed by all basins in the network, and passed on to the label prediction model, which calculates the water level (and the loss function). The output from this iteration of the network is then passed on to inform downstream models, and so on.

An illustration of the HydroNets architecture.

We’re incredibly excited about this progress, and are working hard on improving our systems further.

Acknowledgements
This work is a collaboration between many research teams at Google, and is part of our AI for Social Good efforts. We'd also like to thank our Geo and Policy teams, as well as Google.org.

Source: Google AI Blog


KeyPose: Estimating the 3D Pose of Transparent Objects from Stereo

Estimating the position and orientation of 3D objects is one of the core problems in computer vision applications that involve object-level perception, such as augmented reality and robotic manipulation. In these applications, it is important to know the 3D position of objects in the world, either to directly affect them, or to place simulated objects correctly around them. While there has been much research on this topic using machine learning (ML) techniques, especially Deep Nets, most have relied on the use of depth sensing devices, such as the Kinect, which give direct measurements of the distance to an object. For objects that are shiny or transparent, direct depth sensing does not work well. For example, the figure below includes a number of objects (left), two of which are transparent stars. A depth device does not find good depth values for the stars, and gives a very poor reconstruction of the actual 3D points (right).

Left: RGB image of transparent objects.  Right: A four-panel image showing the reconstructed depth for the scene on the left.The top row includes depth images and the bottom row presents the 3D point cloud. The left panels were reconstructed using a depth camera and the right panels are output from the ClearGrasp model.  Note that although ClearGrasp inpaints the depth of the stars, it mistakes the actual depth of the rightmost one.

One solution to this problem, such as that proposed by ClearGrasp, is to use a deep neural network to inpaint the corrupted depth map of the transparent objects. Given a single RGB-D image of transparent objects, ClearGrasp uses deep convolutional networks to infer surface normals, masks of transparent surfaces, and occlusion boundaries, which it uses to refine the initial depth estimates for all transparent surfaces in the scene (far right in the figure above). This approach is very promising, and allows scenes with transparent objects to be processed by pose-estimation methods that rely on depth.  But inpainting can be tricky, especially when trained completely with synthetic images, and can still result in errors in depth.

In “KeyPose: Multi-View 3D Labeling and Keypoint Estimation for Transparent Objects”, presented at CVPR 2020 in collaboration with the Stanford AI Lab, we describe an ML system that estimates the depth of transparent objects by directly predicting 3D keypoints. To train the system we gather a large real-world dataset of images of transparent objects in a semi-automated way, and efficiently label their pose using 3D keypoints selected by hand. We then train deep models (called KeyPose) to estimate the 3D keypoints end-to-end from monocular or stereo images, without explicitly computing depth. The models work on objects both seen and unseen during training, for both individual objects and categories of objects. While KeyPose can work with monocular images, the extra information available from stereo images allows it to improve its results by a factor of two over monocular image input, with typical errors from 5 mm to 10 mm, depending on the objects. It substantially improves over state-of-the-art in pose estimation for these objects, even when competing methods are provided with ground truth depth. We are releasing the dataset of keypoint-labeled transparent objects for use by the research community.

Real-World Transparent Object Dataset with 3D Keypoint Labels
To facilitate gathering large quantities of real-world images, we set up a robotic data-gathering system in which a robot arm moves through a trajectory while taking video with two devices, a stereo camera and the Kinect Azure depth camera.

Automated image sequence capture using a robot arm with a stereo camera and an Azure Kinect device.

The AprilTags on the target enable accurate tracing of the pose of the cameras. By hand-labelling only a few images in each video with 2D keypoints, we can extract 3D keypoints for all frames of the video using multi-view geometry, thus increasing the labelling efficiency by a factor of 100.

We captured imagery for 15 different transparent objects in five categories, using 10 different background textures and four different poses for each object, yielding a total of 600 video sequences comprising 48k stereo and depth images. We also captured the same images with an opaque version of the object, to provide accurate ground truth depth images. All the images are labelled with 3D keypoints. We are releasing this dataset of real-world images publicly, complementing the synthetic ClearGrasp dataset with which it shares similar objects.

KeyPose Algorithm Using Early Fusion Stereo
The idea of using stereo images directly for keypoint estimation was developed independently for this project; it has also appeared recently in the context of hand-tracking. The diagram below shows the basic idea: the two images from a stereo camera are cropped around the object and fed to the KeyPose network, which predicts a sparse set of 3D keypoints that represent the 3D pose of the object. The network is trained using supervision from the labelled 3D keypoints.

One of the key aspects of stereo KeyPose is the use of early fusion to intermix the stereo images, and allow the network to implicitly compute disparity, in contrast to late fusion, in which keypoints are predicted for each image separately, and then combined. As shown in the diagram below, the output of KeyPose is a 2D keypoint heatmap in the image plane along with a disparity (i.e., inverse depth) heatmap for each keypoint. The combination of these two heatmaps yields the 3D coordinate of the keypoint, for each keypoint.

Keypose system diagram. Stereo images are passed to a CNN model to produce a probability heatmap for each keypoint.  This heatmap yields 2D image coordinates U,V for the keypoint.  The CNN model also produces a disparity (inverse depth) heatmap for each keypoint, which when combined with the U,V coordinates, gives a 3D position (X,Y,Z).

When compared to late fusion or to monocular input, early fusion stereo typically is twice as accurate.

Results
The images below show qualitative results of KeyPose on individual objects. On the left is one of the original stereo images; in the middle are the predicted 3D keypoints projected onto the image. On the right, we visualize points from a 3D model of the bottle, placed at the pose determined by the predicted 3D keypoints. The network is efficient and accurate, predicting keypoints with an MAE of 5.2 mm for the bottle and 10.1 mm for the mug using just 5 ms on a standard GPU.

The following table shows results for KeyPose on category-level estimation. The test set used a background texture not seen by the training set. Note that the MAE varies from 5.8 mm to 9.9 mm, showing the accuracy of the method.

Quantitative comparison of KeyPose with the state-of-the-art DenseFusion system, on category-level data. We provide DenseFusion with two versions of depth, one from the transparent objects, and one from opaque objects. <2cm is the percent of estimates with errors less than 2 cm. MAE is the mean absolute error of the keypoints, in mm.

For a complete accounting of quantitative results, as well as, ablation studies, please see the paper and supplementary materials and the KeyPose website.

Conclusion
This work shows that it is possible to accurately estimate the 3D pose of transparent objects from RGB images without reliance on depth images. It validates the use of stereo images as input to an early fusion deep net, where the network is trained to extract sparse 3D keypoints directly from the stereo pair. We hope the availability of an extensive, labelled dataset of transparent objects will help to advance the field. Finally, while we used semi-automatic methods to efficiently label the dataset, we hope to employ self-supervision methods in future work to do away with manual labelling.

Acknowledgements
I want to thank my co-authors, Xingyu Liu of Stanford University, and Rico Jonschkowski and Anelia Angelova; as well the many who helped us through discussions during the project and paper writing, including Andy Zheng, Shuran Song, Vincent Vanhoucke, Pete Florence, and Jonathan Tompson.

Source: Google AI Blog


Using Machine Learning to Detect Deficient Coverage in Colonoscopy Screenings

Colorectal cancer (CRC) is a global health problem and the second deadliest cancer in the United States, resulting in an estimated 900K deaths per year. While deadly, CRC can be prevented by removing small precancerous lesions in the colon, called polyps, before they become cancerous. In fact, it is estimated that a 1% increase in the adenoma detection rate (ADR, defined as the fraction of procedures in which a physician discovers at least one polyp) can lead to a 6% decrease in the rate of interval CRCs (a CRC that is diagnosed within 60 months of a negative colonoscopy).

Colonoscopy is considered the gold standard procedure for the detection and removal of polyps. Unfortunately, the literature indicates that endoscopists miss on average 22%-28% of polyps during colonoscopies; furthermore, 20% to 24% of polyps that have the potential to become cancerous (adenomas) are missed. Two major factors that may cause an endoscopist to miss a polyp are (1) the polyp appears in the field of view, but the endoscopist misses it, perhaps due to its small size or flat shape; and (2) the polyp does not appear in the field of view, as the endoscopist has not fully covered the relevant area during the procedure.

In “Detecting Deficient Coverage in Colonoscopies”, we introduce the Colonoscopy Coverage Deficiency via Depth algorithm, or C2D2, a machine learning-based approach to improving colonoscopy coverage. The C2D2 algorithm performs a local 3D reconstruction of the colon as images are captured during the procedure, and on that basis, identifies which areas of the colon were covered and which remained outside of the field of view. C2D2 can then indicate in real time whether a particular area of the colon has suffered from deficient coverage so the endoscopist can return to that area. Our work proposes a novel approach to compute coverage in real time, for which 3D reconstruction is done using a calibration-free, unsupervised learning method, and evaluate it in a large scale way.

The C2D2 Algorithm
When considering colon coverage, it is important to estimate the coverage fraction — what percentage of the relevant regions were covered by a complete procedure. While a retrospective analysis is useful for the physician and could provide general guidance for future procedures, it is more useful to have real-time estimation of coverage fraction, on a segment by segment basis, i.e. knowledge of what fraction of the current segment has been covered while traversing the colon. The helpfulness of such functionality is clear: during the procedure itself, a physician may be alerted to segments with deficient coverage, and can immediately return to review these areas. Higher coverage will result in a higher proportion of polyps being seen.

The C2D2 algorithm is designed to compute such a segment-by-segment coverage in two phases: computing depth maps for each frame of the colonoscopy video, followed by computation of coverage based on these depth maps.

C2D2 computes a depth image from a single RGB image. Then, based on the computed depth images for a video sequence, C2D2 calculates local coverage, so it can detect where the coverage has been deficient and a second look is required.

Depth map creation consists of both depth estimation as well as pose estimation — the localization of where the endoscope is in space, as well as the direction it is pointing. In addition to the detection of deficient coverage, depth and pose estimation are useful for a variety of other interesting tasks. For example, depth can be used for improved detection of flat polyps, while pose estimation can be used for relocalizing areas of the colon (including polyps) that the endoscopist wishes to revisit, and both together can be used for visualization and navigation.

Top row: RGB image, from which the depth is computed. Bottom row: Depth image as computed by C2D2. Yellow is deeper, blue is shallower. Note that the “tunnel” structure is captured, as well as the Haustral ridges.

In order to compute coverage fractions from these depth maps, we trained C2D2 on two sources of data: synthetic sequences and real sequences. We generated the synthetic videos using a graphical model of a colon. For each synthetic video, ground truth coverage is available in the form of a number between 0 (completely uncovered) and 1 (completely covered). For real sequences, we analyzed de-identified colonoscopy videos, for which ground truth coverage is unavailable.

Performance on Synthetic Videos
When using synthetic videos, the availability of ground truth coverage enables the direct measurement of C2D2’s performance. We quantify this using the mean absolute error (MAE), which indicates how much the algorithm’s prediction differs, on average, from the ground truth. We find that C2D2’s MAE = 0.075; meaning that, on average, the prediction of C2D2 is within 7.5% of the ground truth. By contrast, a group of physicians given the same task achieved MAE = 0.177, i.e., within 17.7% of the ground truth. Thus, the C2D2 attained an accuracy rate 2.4 times higher on synthetic sequences.

Performance on Real Videos
Of course, what matters most is performance on videos of real colonoscopies. The challenge in this case is the absence of ground truth labelling: we don’t know what the actual coverage is. Additionally, one cannot use labels provided by experts directly as they are not always accurate, due to the challenges described earlier. However, C2D2 can still perform inference on real colonoscopy videos. Indeed, the learning pipeline is designed to perform equally well on synthetic and real colonoscopy videos.

To verify performance on real sequences, we used a variant of a technique common in the generative modelling literature, which involves providing video sequences to human experts along with C2D2’s coverage scores for those sequences. We then ask the experts to assess whether C2D2’s score is correct. The idea is that while it is difficult for experts to assign a score directly, the task of verifying a given score is considerably easier. (This is similar to the fact that verifying a proposed solution to an algorithmic problem is generally much easier than computing that solution.) Using this methodology, experts verified C2D2’s score 93% of the time. And in a more qualitative sense, C2D2’s output seems to pass the “eyeball test”, see the figure below.

Coverage on real colonoscopy sequences. Top row: Frames from a well covered sequence — the entire “tunnel” down the lumen may be seen; C2D2 coverage = 0.931. Middle row: A partially covered sequence — the bottom may be seen, but the top is not as visible; C2D2 coverage = 0.427. Bottom row: A poorly covered sequence, much of what is seen is the wall; C2D2 coverage = 0.227.

Next steps
By alerting physicians to missed regions of the colon wall, C2D2 promises to lead to the discovery of more adenomas, thereby increasing the ADR and concomitantly decreasing the rate of interval CRC. This would be of tremendous benefit to patients.

In addition to this work that addresses colonoscopy coverage, we are concurrently conducting research to improve polyp detection by combining C2D2 with an automatic, real-time polyp detection algorithm. This study adds to the mounting evidence that physicians may use machine learning methods to augment their efforts, especially during procedures, to improve the quality of care for patients.

Acknowledgements
This research was conducted by Daniel Freedman, Yochai Blau, Liran Katzir, Amit Aides, Ilan Shimshoni, Danny Veikherman, Tomer Golany, Ariel Gordon, Greg Corrado, Yossi Matias, and Ehud Rivlin, with support from Verily. We would like to thank all of our team members and collaborators who worked on this project with us, including: Nadav Rabani, Chen Barshai, Nia Stoykova, David Ben-Shimol, Jesse Lachter, and Ori Segol, 3D-Systems and many others. We'd also like to thank Yossi Matias for support and guidance. The research was conducted by teams from Google Health and Google Research, Israel.

Source: Google AI Blog


Scaling Up Fundamental Quantum Chemistry Simulations on Quantum Hardware

Accurate computational prediction of chemical processes from the quantum mechanical laws that govern them is a tool that can unlock new frontiers in chemistry, improving a wide variety of industries. Unfortunately, the exact solution of quantum chemical equations for all but the smallest systems remains out of reach for modern classical computers, due to the exponential scaling in the number and statistics of quantum variables. However, by using a quantum computer, which by its very nature takes advantage of unique quantum mechanical properties to handle calculations intractable to its classical counterpart, simulations of complex chemical processes can be achieved. While today’s quantum computers are powerful enough for a clear computational advantage at some tasks, it is an open question whether such devices can be used to accelerate our current quantum chemistry simulation techniques.

In “Hartree-Fock on a Superconducting Qubit Quantum Computer”, appearing today in Science, the Google AI Quantum team explores this complex question by performing the largest chemical simulation performed on a quantum computer to date. In our experiment, we used a noise-robust variational quantum eigensolver (VQE) to directly simulate a chemical mechanism via a quantum algorithm. Though the calculation focused on the Hartree-Fock approximation of a real chemical system, it was twice as large as previous chemistry calculations on a quantum computer, and contained ten times as many quantum gate operations. Importantly, we validate that algorithms being developed for currently available quantum computers can achieve the precision required for experimental predictions, revealing pathways towards realistic simulations of quantum chemical systems. Furthermore, we have released the code for the experiment, which uses OpenFermion, our open source repository for quantum computations of chemistry.

Google’s Sycamore processor mounted in a cryostat, recently used to demonstrate quantum supremacy and the largest quantum chemistry simulation on a quantum computer. Photo Credit: Rocco Ceselin

Developing an Error Robust Quantum Algorithm for Chemistry
There are a number of ways to use a quantum computer to simulate the ground state energy of a molecular system. In this work we focused on a quantum algorithm “building block”, or circuit primitive, and perfect its performance through a VQE (more on that later). In the classical setting this circuit primitive is equivalent to the Hartree-Fock model and is an important circuit component of an algorithm we previously developed for optimal chemistry simulations. This allows us to focus on scaling up without incurring exponential simulation costs to validate our device. Therefore, robust error mitigation on this component is crucial for accurate simulations when scaling to the “beyond classical” regime.

Errors in quantum computation emerge from interactions of the quantum circuitry with the environment, causing erroneous logic operations — even minor temperature fluctuations can cause qubit errors. Algorithms for simulating chemistry on near-term quantum devices must account for these errors with low overhead, both in terms of the number of qubits or additional quantum resources, such as implementing a quantum error correcting code. The most popular method to account for errors (and why we used it for our experiment) is to use a VQE. For our experiment, we selected the VQE we developed a few years ago, which treats the quantum processor like an neural network and attempts to optimize a quantum circuit’s parameters to account for noisy quantum logic by minimizing a cost function. Just like how classical neural networks can tolerate imperfections in data by optimization, a VQE dynamically adjusts quantum circuit parameters to account for errors that occur during the quantum computation.

Enabling High Accuracy with Sycamore
The experiment was run on the Sycamore processor that was recently used to demonstrate quantum supremacy. Though our experiment required fewer qubits, even higher quantum gate fidelity was needed to resolve chemical bonding. This led to the development of new, targeted calibration techniques that optimally amplify errors so they can be diagnosed and corrected.

Energy predictions of molecular geometries by the Hartree-Fock model simulated on 10 qubits of the Sycamore processor.

Errors in the quantum computation can originate from a variety of sources in the quantum hardware stack. Sycamore has 54-qubits and consists of over 140 individually tunable elements, each controlled with high-speed, analog electrical pulses. Achieving precise control over the whole device requires fine tuning more than 2,000 control parameters, and even small errors in these parameters can quickly add up to large errors in the total computation.

To accurately control the device, we use an automated framework that maps the control problem onto a graph with thousands of nodes, each of which represent a physics experiment to determine a single unknown parameter. Traversing this graph takes us from basic priors about the device to a high fidelity quantum processor, and can be done in less than a day. Ultimately, these techniques along with the algorithmic error mitigation enabled orders of magnitude reduction in the errors.

Left: The energy of a linear chain of Hydrogen atoms as the bond distance between each atom is increased. The solid line is the Hartree-Fock simulation with a classical computer while the points are computed with the Sycamore processor. Right: Two accuracy metrics (infidelity and mean absolute error) for each point computed with Sycamore. “Raw” is the non-error-mitigated data from Sycamore. “+PS” is data from a type of error mitigation correcting the number of electrons. “+Purification” is a type of error mitigation correcting for the right kind of state. “+VQE” is the combination of all the error mitigation along with variational relaxation of the circuit parameters. Experiments on H8, H10, and H12 show similar performance improvements upon error mitigation.

Pathways Forward
We hope that this experiment serves as a blueprint for how to run chemistry calculations on quantum processors, and as a jumping off point on the path to physical simulation advantage. One exciting prospect is that it is known how to modify the quantum circuits used in this experiment in a simple way such that they are no longer efficiently simulable, which would determine new directions for improved quantum algorithms and applications. We hope that the results from this experiment can be used to explore this regime by the broader research community. To run these experiments, you can find the code here.

Source: Google AI Blog