Video conferencing is becoming ever more critical in people's work and personal lives. Improving that experience with privacy enhancements or fun visual touches can help center our focus on the meeting itself. As part of this goal, we recently announced ways to blur and replace your background in Google Meet, which use machine learning (ML) to better highlight participants regardless of their surroundings. Whereas other solutions require installing additional software, Meet’s features are powered by cutting-edge web ML technologies built with MediaPipe that work directly in your browser — no extra steps necessary. One key goal in developing these features was to provide real-time, in-browser performance on almost all modern devices, which we accomplished by combining efficient on-device ML models, WebGL-based rendering, and web-based ML inference via XNNPACK and TFLite.
Overview of Our Web ML Solution The new features in Meet are developed with MediaPipe, Google's open source framework for cross-platform customizable ML solutions for live and streaming media, which also powers ML solutions like on-device real-time hand, iris and body pose tracking.
A core need for any on-device solution is to achieve high performance. To accomplish this, MediaPipe’s web pipeline leverages WebAssembly, a low-level binary code format designed specifically for web browsers that improves speed for compute-heavy tasks. At runtime, the browser converts WebAssembly instructions into native machine code that executes much faster than traditional JavaScript code. In addition, Chrome 84 recently introduced support for WebAssembly SIMD, which processes multiple data points with each instruction, resulting in a performance boost of more than 2x.
Our solution first processes each video frame by segmenting a user from their background (more about our segmentation model later in the post) utilizing ML inference to compute a low resolution mask. Optionally, we further refine the mask to align it with the image boundaries. The mask is then used to render the video output via WebGL2, with the background blurred or replaced.
In the current version, model inference is executed on the client’s CPU for low power consumption and widest device coverage. To achieve real-time performance, we designed efficient ML models with inference accelerated by the XNNPACK library, the first inference engine specifically designed for the novel WebAssembly SIMD specification. Accelerated by XNNPACK and SIMD, the segmentation model can run in real-time on the web.
Enabled by MediaPipe's flexible configuration, the background blur/replace solution adapts its processing based on device capability. On high-end devices it runs the full pipeline to deliver the highest visual quality, whereas on low-end devices it continues to perform at speed by switching to compute-light ML models and bypassing the mask refinement.
Segmentation Model On-device ML models need to be ultra lightweight for fast inference, low power consumption, and small download size. For models running in the browser, the input resolution greatly affects the number of floating-point operations (FLOPs) necessary to process each frame, and therefore needs to be small as well. We downsample the image to a smaller size before feeding it to the model. Recovering a segmentation mask as fine as possible from a low-resolution image adds to the challenges of model design.
The overall segmentation network has a symmetric structure with respect to encoding and decoding, while the decoder blocks (light green) also share a symmetric layer structure with the encoder blocks (light blue). Specifically, channel-wise attention with global average pooling is applied in both encoder and decoder blocks, which is friendly to efficient CPU inference.
We modified MobileNetV3-small as the encoder, which has been tuned by network architecture search for the best performance with low resource requirements. To reduce the model size by 50%, we exported our model to TFLite using float16 quantization, resulting in a slight loss in weight precision but with no noticeable effect on quality. The resulting model has 193K parameters and is only 400KB in size.
Rendering Effects Once segmentation is complete, we use OpenGL shaders for video processing and effect rendering, where the challenge is to render efficiently without introducing artifacts. In the refinement stage, we apply a joint bilateral filter to smooth the low resolution mask.
The blur shader simulates a bokeh effect by adjusting the blur strength at each pixel proportionally to the segmentation mask values, similar to the circle-of-confusion (CoC) in optics. Pixels are weighted by their CoC radii, so that foreground pixels will not bleed into the background. We implemented separable filters for the weighted blur, instead of the popular Gaussian pyramid, as it removes halo artifacts surrounding the person. The blur is performed at a low resolution for efficiency, and blended with the input frame at the original resolution.
For background replacement, we adopt a compositing technique, known as light wrapping, for blending segmented persons and customized background images. Light wrapping helps soften segmentation edges by allowing background light to spill over onto foreground elements, making the compositing more immersive. It also helps minimize halo artifacts when there is a large contrast between the foreground and the replaced background.
Performance To optimize the experience for different devices, we provide model variants at multiple input sizes (i.e., 256x144 and 160x96 in the current release), automatically selecting the best according to available hardware resources.
We evaluated the speed of model inference and the end-to-end pipeline on two common devices: MacBook Pro 2018 with 2.2 GHz 6-Core Intel Core i7, and Acer Chromebook 11 with Intel Celeron N3060. For 720p input, the MacBook Pro can run the higher-quality model at 120 FPS and the end-to-end pipeline at 70 FPS, while the Chromebook runs inference at 62 FPS with the lower-quality model and 33 FPS end-to-end.
For quantitative evaluation of model accuracy, we adopt the popular metrics of intersection-over-union (IOU) and boundary F-measure. Both models achieve high quality, especially for having such a lightweight network:
We also release the accompanying Model Card for our segmentation models, which details our fairness evaluations. Our evaluation data contains images from 17 geographical subregions of the globe, with annotations for skin tone and gender. Our analysis shows that the model is consistent in its performance across the various regions, skin-tones, and genders, with only small deviations in IOU metrics.
Conclusion We introduced a new in-browser ML solution for blurring and replacing your background in Google Meet. With this, ML models and OpenGL shaders can run efficiently on the web. The developed features achieve real-time performance with low power consumption, even on low-power devices.
Acknowledgments Special thanks to those on the Meet team and others who worked on this project, in particular Sebastian Jansson, Rikard Lundmark, Stephan Reiter, Fabian Bergmark, Ben Wagner, Stefan Holmer, Dan Gunnarson, Stéphane Hulaud and to all our team members who worked on the technology with us: Siargey Pisarchyk, Karthik Raveendran, Chris McClanahan, Marat Dukhan, Frank Barchard, Ming Guang Yong, Chuo-Ling Chang, Michael Hays, Camillo Lugaresi, Gregory Karpiak, Siarhei Kazakou, Matsvei Zhdanovich, and Matthias Grundmann.
At Google, we're actively exploring how people can use creativity tools powered by machine learning and computational methods when producing multimedia content, from creating music and reframing videos, to drawing and more. One creative process in particular, video production, can especially benefit from such tools, as it requires a series of decisions about what content is best suited to a target audience, how to position the available assets within the field of view, and what temporal arrangement will yield the most compelling narrative. But what if one could leverage existing assets, such as a website, to get a jump-start on video creation? Businesses commonly host websites that contain rich visual representations about their services or products, all of which could be repurposed for other multimedia formats, such as videos, potentially enabling those without extensive resources the ability to reach a broader audience.
In “Automatic Video Creation From a Web Page”, published at UIST 2020, we introduce URL2Video, a research prototype pipeline to automatically convert a web page into a short video, given temporal and visual constraints provided by the content owner. URL2Video extracts assets (text, images, or videos) and their design styles (including fonts, colors, graphical layouts, and hierarchy) from HTML sources and organizes the visual assets into a sequence of shots, while maintaining a look-and-feel similar to the source page. Given a user-specified aspect ratio and duration, it then renders the repurposed materials into a video that is ideal for product and service advertising.
URL2Video Overview Assume a user provides an URL to a web page that illustrates their business. The URL2Video pipeline automatically selects key content from the page and decides the temporal and visual presentation of each asset, based on a set of heuristics derived from an interview study with designers who were familiar with web design and video ad creation. These designer-informed heuristics capture common video editing styles, including content hierarchy, constraining the amount of information in a shot and its time duration, providing consistent color and style for branding, and more. Using this information, the URL2Video pipeline parses a web page, analyzing the content and selecting visually salient text or images while preserving their design styles, which it organizes according to the video specifications provided by the user.
Webpage Analysis Given a webpage URL, URL2Video extracts document object model (DOM) information and multimedia materials. For the purposes of our research prototype, we limited the domain to static web pages that contain salient assets and headings preserved in an HTML hierarchy that follows recent web design principles, which encourage the use of prominent elements, distinct sections, and an order of visual focus that guides readers in perceiving information. URL2Video identifies such visually-distinguishable elements as a candidate list of asset groups, each of which may contain a heading, a product image, detailed descriptions, and call-to-action buttons, and captures both the raw assets (text and multimedia files) and detailed design specifications (HTML tags, CSS styles, and rendered locations) for each element. It then ranks the asset groups by assigning each a priority score based on their visual appearance and annotations, including their HTML tags, rendered sizes, and ordering shown on the page. In this way, an asset group that occupies a larger area at the top of the page receives a higher score.
Constraints-Based Asset Selection We consider two goals when composing a video: (1) each video shot should provide concise information, and (2) the visual design should be consistent with the source page. Based on these goals and the video constraints provided by the user, including the intended video duration (in seconds) and aspect ratio (commonly 16:9, 4:3, 1:1, etc.), URL2Video automatically selects and orders the asset groups to optimize the total priority score. To make the content concise, it presents only dominant elements from a page, such as a headline and a few multimedia assets. It constrains the duration of each visual element for viewers to perceive the content. In this way, a short video highlights the most salient information from the top of the page, and a longer video contains more campaigns or products.
Scene Composition & Video Rendering Given an ordered list of assets based on the DOM hierarchy, URL2Video follows the design heuristics obtained from interview studies to make decisions about both the temporal and spatial arrangement to present the assets in individual shots. It transfers the graphical layout of elements into the video’s aspect ratio, and applies the style choices including fonts and colors. To make a video more dynamic and engaging, it adjusts the presentation timing of assets. Finally, it renders the content into a video in the MPEG-4 container format.
User Control The interface to the research prototype allows the user to review the design attributes in each video shot extracted from the source page, reorder the materials, change the detailed design, such as colors and fonts, and adjust the constraints to generate a new video.
URL2Video Use Cases We demonstrate the performance of the end-to-end URL2Video pipeline on a variety of existing web pages. Below we highlight an example result where URL2Video converts a page that embeds multiple short video clips into a 12-second output video. Note how the pipeline makes automatic editing decisions on font and color choices, timing, and content ordering in a video captured from the source page.
The video below provides further demonstration:
To evaluate the automatically-generated videos, we conducted a user study with designers at Google. Our results show that URL2Video effectively extracted design elements from a web page and supported designers by bootstrapping the video creation process.
Next steps While this current research focuses on the visual presentation, we are developing new techniques that support the audio track and a voiceover in video editing. All in all, we envision a future where creators focus on making high-level decisions and an ML model interactively suggests detailed temporal and graphical edits for a final video creation on multiple platforms.
Acknowledgments We greatly thank our paper co-authors, Zheng Sun (Research) and Katrina Panovich (YouTube). We would also like to thank our colleagues who contributed to URL2Video, (in alphabetical order of last name) Jordan Canedy, Brian Curless, Nathan Frey, Madison Le, Alireza Mahdian, Justin Parra, Emily Ryan, Mogan Shieh, Sandor Szego, and Weilong Yang. We are grateful to receive the support from our leadership, Tomas Izo, Rahul Sukthankar, and Jay Yagnik.
Recent work suggests that not all data samples are equally useful for training, particularly for deep neural networks (DNNs). Indeed, if a dataset contains low-quality or incorrectly labeled data, one can often improve performance by removing a significant portion of training samples. Moreover, in cases where there is a mismatch between the train and test datasets (e.g., due to difference in train and test location or time), one can also achieve higher performance by carefully restricting samples in the training set to those most relevant for the test scenario. Because of the ubiquity of these scenarios, accurately quantifying the values of training samples has great potential for improving model performance on real-world datasets.
In addition to improving model performance, assigning a quality value to individual data can also enable new use cases. It can be used to suggest better practices for data collection, e.g., what kinds of additional data would benefit the most, and can be used to construct large-scale training datasets more efficiently, e.g., by web searching using the labels as keywords and filtering out less valuable data.
In “Data Valuation Using Deep Reinforcement Learning”, accepted at ICML 2020, we address the challenge of quantifying the value of training data using a novel approach based on meta-learning. Our method integrates data valuation into the training procedure of a predictor model that learns to recognize samples that are more valuable for the given task, improving both predictor and data valuation performance. We have also launched four AI Hub Notebooks that exemplify the use cases of DVRL and are designed to be conveniently adapted to other tasks and datasets, such as domain adaptation, corrupted sample discovery and robust learning, transfer learning on image data and data valuation.
Quantifying the Value of Data Not all data are equal for a given ML model — some have greater relevance for the task at hand or are more rich in informative content than others. So how does one evaluate the value of a single datum? At the granularity of a full dataset, it is straightforward; one can simply train a model on the entire dataset and use its performance on a test set as its value. However, estimating the value of a single datum is far more difficult, especially for complex models that rely on large-scale datasets, because it is computationally infeasible to re-train and re-evaluate a model on all possible subsets.
To tackle this, researchers have explored permutation-based methods (e.g., influence functions), and game theory-based methods (e.g., data Shapley). However, even the best current methods are far from being computationally feasible for large datasets and complex models, and their data valuation performance is limited. Concurrently, meta learning-based adaptive weight assignment approaches have been developed to estimate the weight values using a meta-objective. But rather than prioritizing learning from high value data samples, their data value mapping is typically based on gradient descent learning or other heuristic approaches that alter the conventional predictor model training dynamics, which can result in performance changes that are unrelated to the value of individual data points.
Data Valuation Using Reinforcement Learning (DVRL) To infer the data values, we propose a data value estimator (DVE) that estimates data values and selects the most valuable samples to train the predictor model. This selection operation is fundamentally non-differentiable and thus conventional gradient descent-based methods cannot be used. Instead, we propose to use reinforcement learning (RL) such that the supervision of the DVE is based on a reward that quantifies the predictor performance on a small (but clean) validation set. The reward guides the optimization of the policy towards the action of optimal data valuation, given the state and input samples. Here, we treat the predictor model learning and evaluation framework as the environment, a novel application scenario of RL-assisted machine learning.
Results We evaluate the data value estimation quality of DVRL on multiple types of datasets and use cases.
Conclusions We propose a novel meta learning framework for data valuation which determines how likely each training sample will be used in training of the predictor model. Unlike previous works, our method integrates data valuation into the training procedure of the predictor model, allowing the predictor and DVE to improve each other's performance. We model this data value estimation task using a DNN trained through RL with a reward obtained from a small validation set that represents the target task performance. In a computationally-efficient way, DVRL can provide high quality ranking of training data that is useful for domain adaptation, corrupted sample discovery and robust learning. We show that DVRL significantly outperforms alternative methods on diverse types of tasks and datasets.
Acknowledgements We gratefully acknowledge the contributions of Tomas Pfister.
Transformer models have achieved state-of-the-art results across a diverse range of domains, including natural language, conversation, images, and even music. The core block of every Transformer architecture is the attention module, which computes similarity scores for all pairs of positions in an input sequence. This however, scales poorly with the length of the input sequence, requiring quadratic computation time to produce all similarity scores, as well as quadratic memory size to construct a matrix to store these scores.
For applications where long-range attention is needed, several fast and more space-efficient proxies have been proposed such as memory caching techniques, but a far more common way is to rely on sparse attention. Sparse attention reduces computation time and the memory requirements of the attention mechanism by computing a limited selection of similarity scores from a sequence rather than all possible pairs, resulting in a sparse matrix rather than a full matrix. These sparse entries may be manually proposed, found via optimization methods, learned, or even randomized, as demonstrated by such methods as Sparse Transformers, Longformers, Routing Transformers, Reformers, and Big Bird. Since sparse matrices can also be represented by graphs and edges, sparsification methods are also motivated by the graph neural network literature, with specific relationships to attention outlined in Graph Attention Networks. Such sparsity-based architectures usually require additional layers to implicitly produce a full attention mechanism.
Unfortunately, sparse attention methods can still suffer from a number of limitations. (1) They require efficient sparse-matrix multiplication operations, which are not available on all accelerators; (2) they usually do not provide rigorous theoretical guarantees for their representation power; (3) they are optimized primarily for Transformer models and generative pre-training; and (4) they usually stack more attention layers to compensate for sparse representations, making them difficult to use with other pre-trained models, thus requiring retraining and significant energy consumption. In addition to these shortcomings, sparse attention mechanisms are often still not sufficient to address the full range of problems to which regular attention methods are applied, such as Pointer Networks. There are also some operations that cannot be sparsified, such as the commonly used softmax operation, which normalizes similarity scores in the attention mechanism and is used heavily in industry-scale recommender systems.
To resolve these issues, we introduce the Performer, a Transformer architecture with attention mechanisms that scale linearly, thus enabling faster training while allowing the model to process longer lengths, as required for certain image datasets such as ImageNet64 and text datasets such as PG-19. The Performer uses an efficient (linear) generalized attention framework, which allows a broad class of attention mechanisms based on different similarity measures (kernels). The framework is implemented by our novel Fast Attention Via Positive Orthogonal Random Features (FAVOR+) algorithm, which provides scalable low-variance and unbiased estimation of attention mechanisms that can be expressed by random feature map decompositions (in particular, regular softmax-attention). We obtain strong accuracy guarantees for this method while preserving linear space and time complexity, which can also be applied to standalone softmax operations.
Generalized Attention In the original attention mechanism, the query and key inputs, corresponding respectively to rows and columns of a matrix, are multiplied together and passed through a softmax operation to form an attention matrix, which stores the similarity scores. Note that in this method, one cannot decompose the query-key product back into its original query and key components after passing it into the nonlinear softmax operation. However, it is possible to decompose the attention matrix back to a product of random nonlinear functions of the original queries and keys, otherwise known as random features, which allows one to encode the similarity information in a more efficient manner.
Regular softmax-attention can be seen as a special case with these nonlinear functions defined by exponential functions and Gaussian projections. Note that we can also reason inversely, by implementing more general nonlinear functions first, implicitly defining other types of similarity measures, or kernels, on the query-key product. We frame this as generalized attention, based on earlier work in kernel methods. Although for most kernels, closed-form formulae do not exist, our mechanism can still be applied since it does not rely on them.
To the best of our knowledge, we are the first to show that any attention matrix can be effectively approximated in downstream Transformer-applications using random features. The novel mechanism enabling this is the use of positive random features, i.e., positive-valued nonlinear functions of the original queries and keys, which prove to be crucial for avoiding instabilities during training and provide more accurate approximation of the regular softmax attention mechanism.
Towards FAVOR+: Fast Attention via Matrix Associativity The decomposition described above allows one to store the implicit attention matrix with linear, rather than quadratic, memory complexity. One can also obtain a linear time attention mechanism using this decomposition. While the original attention mechanism multiplies the stored attention matrix with the value input to obtain the final result, after decomposing the attention matrix, one can rearrange matrix multiplications to approximate the result of the regular attention mechanism, without explicitly constructing the quadratic-sized attention matrix. This ultimately leads to FAVOR+.
The above analysis is relevant for so-called bidirectional attention, i.e., non-causal attention where there is no notion of past and future. For unidirectional (causal) attention, where tokens do not attend to other tokens appearing later in the input sequence, we slightly modify the approach to use prefix-sum computations, which only store running totals of matrix computations rather than storing an explicit lower-triangular regular attention matrix.
Properties We first benchmark the space- and time-complexity of the Performer and show that the attention speedups and memory reductions are empirically nearly optimal, i.e., very close to simply not using an attention mechanism at all in the model.
We further show that the Performer, using our unbiased softmax approximation, is backwards compatible with pretrained Transformer models after a bit of fine-tuning, which could potentially lower energy costs by improving inference speed, without having to fully retrain pre-existing models.
Example Application: Protein Modeling Proteins are large molecules with complex 3D structures and specific functions that are essential to life. Like words, proteins are specified as linear sequences where each character is one of 20 amino acid building blocks. Applying Transformers to large unlabeled corpora of protein sequences (e.g. UniRef) yields models that can be used to make accurate predictions about the folded, functional macromolecule. Performer-ReLU (which uses ReLU-based attention, an instance of generalized attention that is different from softmax) performs strongly at modeling protein sequence data, while Performer-Softmax matches the performance of the Transformer, as predicted by our theoretical results.
Below we visualize a protein Performer model, trained using the ReLU-based approximate attention mechanism. Using the Performer to estimate similarity between amino acids recovers similar structure to well-known substitution matrices obtained by analyzing evolutionary substitution patterns across carefully curated sequence alignments. More generally, we find local and global attention patterns consistent with Transformer models trained on protein data. The dense attention approximation of the Performer has the potential to capture global interactions across multiple protein sequences. As a proof of concept, we train models on long concatenated protein sequences, which overloads the memory of a regular Transformer model, but not the Performer due to its space efficiency.
Conclusion Our work contributes to the recent efforts on non-sparsity based methods and kernel-based interpretations of Transformers. Our method is interoperable with other techniques like reversible layers and we have even integrated FAVOR with the Reformer's code. We provide the links for the paper, Performer code, and the Protein Language Modeling code. We believe that our research opens up a brand new way of thinking about attention, Transformer architectures, and even kernel methods.
Acknowledgements This work was performed by the core Performer designers Krzysztof Choromanski (Google Brain Team, Tech and Research Lead), Valerii Likhosherstov (University of Cambridge) and Xingyou Song (Google Brain Team), with contributions from David Dohan, Andreea Gane, Tamas Sarlos, Peter Hawkins, Jared Davis, Afroz Mohiuddin, Lukasz Kaiser, David Belanger, Lucy Colwell, and Adrian Weller. We give special thanks to the Applied Science Team for jointly leading the research effort on applying efficient Transformer architectures to protein sequence data.
We additionally wish to thank Joshua Meier, John Platt, and Tom Weingarten for many fruitful discussions on biological data and useful comments on this draft, along with Yi Tay and Mostafa Dehghani for discussions on comparing baselines. We further thank Nikita Kitaev and Wojciech Gajewski for multiple discussions on the Reformer, and Aurko Roy and Ashish Vaswani for multiple discussions on the Routing Transformer.
At Google, it is our ongoing goal to support faculty who are conducting innovative research that will have positive societal impact. As part of that goal, earlier this year we launched the Award for Inclusion Research program, a global program that supports academic research in computing and technology addressing the needs of underrepresented populations. The Award for Inclusion Research program allows faculty and Google researchers an opportunity to partner on their research initiatives and build new and constructive long-term relationships.
We received 100+ applications from over 100 universities, globally, and today we are excited to announce the 16 proposals chosen for funding, focused on an array of topics around diversity and inclusion, algorithmic bias, education innovation, health tools, accessibility, gender bias, AI for social good, security, and social justice. The proposals include 25 principal investigators who focus on making the community stronger through their research efforts.
Congratulations to announce this year’s recipients:
"Human Centred Technology Design for Social Justice in Africa" Anicia Peters (University of Namibia) and Shaimaa Lazem (City for Scientific Research and Technological Applications, Egypt)
"Modern NLP for Regional and Dialectal Language Variants" Antonios Anastasopoulos (George Mason University)
"Culturally Relevant Collaborative Health Tracking Tools for Motivating Heart-Healthy Behaviors Among African Americans" Aqueasha Martin-Hammond (Indiana University - Purdue University Indianapolis) and Tanjala S. Purnell (Johns Hopkins University)
"Characterizing Energy Equity in the United States" Destenie Nock and Constantine Samaras (Carnegie Mellon University)
"Developing a Dialogue System for a Culturally-Responsive Social Programmable Robot" Erin Walker (University of Pittsburgh) and Leshell Hatley (Coppin State University)
"Eliminating Gender Bias in NLP Beyond English" Hinrich Schuetze (LMU Munich)
"The Ability-Based Design Mobile Toolkit: Enabling Accessible Mobile Interactions through Advanced Sensing and Modeling" Jacob O. Wobbrock (University of Washington)
"Mutual aid and community engagement: Community-based mechanisms against algorithmic bias" Jasmine McNealy (University of Florida)
"Empowering Syrian Girls through Culturally Sensitive Mobile Technology and Media Literacy Karen Elizabeth Fisher (University of Washington) and Yacine Ghamri-Doudane (University of La Rochelle)
"Broadening participation in data science through examining the health, social, and economic impacts of gentrification" Latifa Jackson (Howard University) and Hasan Jackson (Howard University)
"Understanding How Peer and Near Peer Mentors co-Facilitating the Active Learning Process of Introductory Data Structures Within an Immersive Summer Experience Effected Rising Sophomore Computer Science Student Persistence and Preparedness for Careers in Silicon Valley" Legand Burge (Howard University) and Marlon Mejias (University of North Carolina at Charlotte)
"Who is Most Likely to Advocate for this Case? A Machine Learning Approach" Maria De-Arteaga (University of Texas at Austin)
"Contextual Rendering of Equations for Visually Impaired Persons" Meenakshi Balakrishnan (Indian Institute of Technology Delhi, India) and Volker Sorge (University of Birmingham)
"Measuring the Cultural Competence of Computing Students and Faculty Nationwide to Improve Diversity, Equity, and Inclusion" Nicki Washington (Duke University)
"Designing and Building Collaborative Tools for Mixed-Ability Programming Teams" Steve Oney (University of Michigan)
"Iterative Design of a Black Studies Research Computing Initiative through `Flipped Research’" Timothy Sherwood and Sharon Tettegah (University of California, Santa Barbara)
For many, gazing at an old photo of a city can evoke feelings of both nostalgia and wonder — what was it like to walk through Manhattan in the 1940s? How much has the street one grew up on changed? While Google Street View allows people to see what an area looks like in the present day, what if you want to explore how places looked in the past?
To create a rewarding “time travel” experience for both research and entertainment purposes, we are launching a browser-based toolset called rǝ (pronounced as re”turn"), an open source, scalable system running on Google Cloud and Kubernetes that can reconstruct cities from historical maps and photos, which represents an implementation of our suite of open source tools launched earlier this year. Referencing the common prefix meaning again or anew, rǝ is meant to represent the themes of reconstruction, research, recreation and remembering behind this crowdsourced research effort, and consists of three components:
Our goal is for rǝ to become a compendium that allows history enthusiasts to virtually experience historical cities around the world, aids researchers, policy makers and educators, and provides a dose of nostalgia to everyday users.
Crowdsourcing Data from Historical Maps Reconstructing how cities used to look at scale is a challenge — historical image data is more difficult to work with than modern data, as there are far fewer images available and much less metadata captured from the images. To help with this difficulty, the rǝ maps module is a suite of open source tools that work together to create a map server with a time dimension, allowing users to jump back and forth between time periods using a slider. These tools allow users to upload scans of historical print maps, georectify them to match real world coordinates, and then convert them to vector format by tracing their geographic features. These vectorized maps are then served on a tile server and rendered as slippy maps, which lets the user zoom in and pan around.
The entry point of the rǝ maps module is Warper, a web app that allows users to upload historical images of maps and georectify them by finding control points on the historical map and corresponding points on a base map. The next app, Editor, allows users to load the georectified historical maps as the background and then trace their geographic features (e.g., building footprints, roads, etc.). This traced data is stored in an OpenStreetMap (OSM) vector format. They are then converted to vector tiles and served from the Server app, a vector tile server. Finally, our map renderer, Kartta, visualizes the spatiotemporal vector tiles allowing the users to navigate space and time on historical maps. These tools were built on top of numerous open source resources including OpenStreetMap, and we intend for our tools and data to be completely open source as well.
3D Experience The 3D Models module aims to reconstruct the detailed full 3D structures of historical buildings using the associated images and maps data, organize these 3D models properly in one repository, and render them on the historical maps with a time dimension.
In many cases, there is only one historical image available for a building, which makes the 3D reconstruction an extremely challenging problem. To tackle this challenge, we developed a coarse-to-fine reconstruction-by-recognition algorithm.
Starting with footprints on maps and façade regions in historical images (both are annotated by crowdsourcing or detected by automatic algorithms), the footprint of one input building is extruded upwards to generate its coarse 3D structure. The height of this extrusion is set to the number of floors from the corresponding metadata in the maps database.
In parallel, instead of directly inferring the detailed 3D structures of each façade as one entity, the 3D reconstruction pipeline recognizes all individual constituent components (e.g., windows, entries, stairs, etc.) and reconstructs their 3D structures separately based on their categories. Then these detailed 3D structures are merged with the coarse one for the final 3D mesh. The results are stored in a 3D repository and ready for 3D rendering.
The key technology powering this feature is a number of state-of-art deep learning models:
Key Results
Conclusion With rǝ, we have developed tools that facilitate crowdsourcing to tackle the main challenge of insufficient historical data when recreating virtual cities. The 3D experience is still a work-in-progress and we aim to improve it with future updates. We hope rǝ acts as a nexus for an active community of enthusiasts and casual users that not only utilizes our historical datasets and open source code, but actively contributes to both.
Acknowledgements This effort has been successful thanks to the hard work of many people, including, but not limited to the following (in alphabetical order of last name): Yale Cong, Feng Han, Amol Kapoor, Raimondas Kiveris, Brandon Mayer, Mark Phillips, Sasan Tavakkol, and Tim Waters (Waters Geospatial Ltd).
Natural language processing (NLP) has seen significant progress over the past several years, with pre-trained models like BERT, ALBERT, ELECTRA, and XLNet achieving remarkable accuracy across a variety of tasks. In pre-training, representations are learned from a large text corpus, e.g., Wikipedia, by repeatedly masking out words and trying to predict them (this is called masked language modeling). The resulting representations encode rich information about language and correlations between concepts, such as surgeons and scalpels. There is then a second training stage, fine-tuning, in which the model uses task-specific training data to learn how to use the general pre-trained representations to do a concrete task, like classification. Given the broad adoption of these representations in many NLP tasks, it is crucial to understand the information encoded in them and how any learned correlations affect performance downstream, to ensure the application of these models aligns with our AI Principles.
In “Measuring and Reducing Gendered Correlations in Pre-trained Models” we perform a case study on BERT and its low-memory counterpart ALBERT, looking at correlations related to gender, and formulate a series of best practices for using pre-trained language models. We present experimental results over public model checkpoints and an academic task dataset to illustrate how the best practices apply, providing a foundation for exploring settings beyond the scope of this case study. We will soon release a series of checkpoints, Zari1, which reduce gendered correlations while maintaining state-of-the-art accuracy on standard NLP task metrics.
Measuring Correlations To understand how correlations in pre-trained representations can affect downstream task performance, we apply a diverse set of evaluation metrics for studying the representation of gender. Here, we’ll discuss results from one of these tests, based on coreference resolution, which is the capability that allows models to understand the correct antecedent to a given pronoun in a sentence. For example, in the sentence that follows, the model should recognize his refers to the nurse, and not to the patient.
The standard academic formulation of the task is the OntoNotes test (Hovy et al., 2006), and we measure how accurate a model is at coreference resolution in a general setting using an F1 score over this data (as in Tenney et al. 2019). Since OntoNotes represents only one data distribution, we also consider the WinoGender benchmark that provides additional, balanced data designed to identify when model associations between gender and profession incorrectly influence coreference resolution. High values of the WinoGender metric (close to one) indicate a model is basing decisions on normative associations between gender and profession (e.g., associating nurse with the female gender and not male). When model decisions have no consistent association between gender and profession, the score is zero, which suggests that decisions are based on some other information, such as sentence structure or semantics.
In this study, we see that neither the (Large) BERT or ALBERT public model achieves zero score on the WinoGender examples, despite achieving impressive accuracy on OntoNotes (close to 100%). At least some of this is due to models preferentially using gendered correlations in reasoning. This isn’t completely surprising: there are a range of cues available to understand text and it is possible for a general model to pick up on any or all of these. However, there is reason for caution, as it is undesirable for a model to make predictions primarily based on gendered correlations learned as priors rather than the evidence available in the input.
Best Practices Given that it is possible for unintended correlations in pre-trained model representations to affect downstream task reasoning, we now ask: what can one do to mitigate any risk this poses when developing new NLP models?
What’s Next We believe these best practices provide a starting point for developing robust NLP systems that perform well across the broadest possible range of linguistic settings and applications. Of course these techniques on their own are not sufficient to capture and remove all potential issues. Any model deployed in a real-world setting should undergo rigorous testing that considers the many ways it will be used, and implement safeguards to ensure alignment with ethical norms, such as Google's AI Principles. We look forward to developments in evaluation frameworks and data that are more expansive and inclusive to cover the many uses of language models and the breadth of people they aim to serve.
Acknowledgements This is joint work with Xuezhi Wang, Ian Tenney, Ellie Pavlick, Alex Beutel, Jilin Chen, Emily Pitler, and Slav Petrov. We benefited greatly throughout the project from discussions with Fernando Pereira, Ed Chi, Dipanjan Das, Vera Axelrod, Jacob Eisenstein, Tulsee Doshi, and James Wexler.
1 Zari is an Afghan Muppet designed to show that ‘a little girl could do as much as everybody else’. [Edit 10/23/2020 — The Zari checkpoints are now available on the GitHub repository.] ↩
Google created the PhD Fellowship Program in 2009 to recognize and support outstanding graduate students who seek to influence the future of technology by pursuing exceptional research in computer science and related fields. Now in its twelfth year, these Fellowships have helped support approximately 500 graduate students globally in North America and Europe, Africa, Australia, East Asia, and India.
It is our ongoing goal to continue to support the academic community as a whole, and these Fellows as they make their mark on the world. We congratulate all of this year’s awardees!
Algorithms, Optimizations and Markets Jan van den Brand, KTH Royal Institute of Technology Mahsa Derakhshan, University of Maryland, College Park Sidhanth Mohanty, University of California, Berkeley
Computational Neuroscience Connor Brennan, University of Pennsylvania
Human Computer Interaction Abdelkareem Bedri, Carnegie Mellon University Brendan David-John, University of Florida Hiromu Yakura, University of Tsukuba Manaswi Saha, University of Washington Muratcan Cicek, University of California, Santa Cruz Prashan Madumal, University of Melbourne
Machine Learning Alon Brutzkus, Tel Aviv University Chin-Wei Huang, Universite de Montreal Eli Sherman, Johns Hopkins University Esther Rolf, University of California, Berkeley Imke Mayer, Fondation Sciences Mathématique de Paris Jean Michel Sarr, Cheikh Anta Diop University Lei Bai, University of New South Wales Nontawat Charoenphakdee, The University of Tokyo Preetum Nakkiran, Harvard University Sravanti Addepalli, Indian Institute of Science Taesik Gong, Korea Advanced Institute of Science and Technology Vihari Piratla, Indian Institute of Technology - Bombay Vishakha Patil, Indian Institute of Science Wilson Tsakane Mongwe, University of Johannesburg Xinshi Chen, Georgia Institute of Technology Yadan Luo, University of Queensland
Machine Perception, Speech Technology and Computer Vision Benjamin van Niekerk, University of Stellenbosch Eric Heiden, University of Southern California Gyeongsik Moon, Seoul National University Hou-Ning Hu, National Tsing Hua University Nan Wu, New York University Shaoshuai Shi, The Chinese University of Hong Kong Yaman Kumar, Indraprastha Institute of Information Technology - Delhi Yifan Liu, University of Adelaide Yu Wu, University of Technology Sydney Zhengqi Li, Cornell University
Mobile Computing Xiaofan Zhang, University of Illinois at Urbana-Champaign
Natural Language Processing Anjalie Field, Carnegie Mellon University Mingda Chen, Toyota Technological Institute at Chicago Shang-Yu Su, National Taiwan University Yanai Elazar, Bar-Ilan
Privacy and Security Julien Gamba, Universidad Carlos III de Madrid Shuwen Deng, Yale University Yunusa Simpa Abdulsalm, Mohammed VI Polytechnic University
Programming Technology and Software Engineering Adriana Sejfia, University of Southern California John Cyphert, University of Wisconsin-Madison
Quantum Computing Amira Abbas, University of KwaZulu-Natal Mozafari Ghoraba Fereshte, EPFL
Structured Data and Database Management Yanqing Peng, University of Utah
Systems and Networking Huynh Nguyen Van, University of Technology Sydney Michael Sammler, Saarland University, MPI-SWS Sihang Liu, University of Virginia Yun-Zhan Cai, National Cheng Kung University
In the last decade, reinforcement learning (RL) has become one of the most promising research areas in machine learning and has demonstrated great potential for solving sophisticated real-world problems, such as chip placement and resource management, and solving challenging games (e.g., Go, Dota 2, and hide-and-seek). In simplest terms, an RL infrastructure is a loop of data collection and training, where actors explore the environment and collect samples, which are then sent to the learners to train and update the model. Most current RL techniques require many iterations over batches of millions of samples from the environment to learn a target task (e.g., Dota 2 learns from batches of 2 million frames every 2 seconds). As such, an RL infrastructure should not only scale efficiently (e.g., increase the number of actors) and collect an immense number of samples, but also be able to swiftly iterate over these extensive amounts of samples during training.
Today we introduce Menger1, a massive large-scale distributed RL infrastructure with localized inference that scales up to several thousand actors across multiple processing clusters (e.g., Borg cells), reducing the overall training time in the task of chip placement. In this post we describe how we implement Menger using Google TPU accelerators for fast training iterations, and present its performance and scalability on the challenging task of chip placement. Menger reduces the training time by up to 8.6x compared to a baseline implementation.
Menger System Design There are various distributed RL systems, such as Acme and SEED RL, each of which focus on optimizing a single particular design point in the space of distributed reinforcement learning systems. For example, while Acme uses local inference on each actor with frequent model retrieval from the learner, SEED RL benefits from a centralized inference design by allocating a portion of TPU cores for performing batched calls. The tradeoffs between these design points are (1) paying the communication cost of sending/receiving observations and actions to/from a centralized inference server or paying the communication cost of model retrieval from a learner and (2) the cost of inference on actors (e.g., CPUs) compared to accelerators (e.g., TPUs/GPUs). Because of the requirements of our target application (e.g., size of observations, actions, and model size), Menger uses local inference in a manner similar to Acme, but pushes the scalability of actors to virtually an unbounded limit. The main challenges to achieving massive scalability and fast training on accelerators include:
Efficient Model Retrieval To address the first challenge, we introduce transparent and distributed caching components between the learner and the actors optimized in TensorFlow and backed by Reverb (similar approach used in Dota). The main responsibility of the caching components is to strike a balance between the large number of requests from actors and the learner job. Adding these caching components not only significantly reduces the pressure on the learner to service the read requests, but also further distributes the actors across multiple Borg cells with a marginal communication overhead. In our study, we show that for a 16 MB model with 512 actors, the introduced caching components reduce the average read latency by a factor of ~4.0x leading to faster training iterations, especially for on-policy algorithms such as PPO.
High Throughput Input Pipeline To deliver a high throughput input data pipeline, Menger uses Reverb, a recently open-sourced data storage system designed for machine learning applications that provides an efficient and flexible platform to implement experience replay in a variety of on-policy/off-policy algorithms. However, using a single Reverb replay buffer service does not currently scale well in a distributed RL setting with thousands of actors, and simply becomes inefficient in terms of write throughput from actors.
To better understand the efficiency of the replay buffer in a distributed setting, we evaluate the average write latency for various payload sizes from 16 MB to 512 MB and a number of actors ranging from 16 to 2048. We repeat the experiment when the replay buffer and actors are placed on the same Borg cell. As the number of actors grows the average write latency also increases significantly. Expanding the number of actors from 16 to 2048, the average write latency increases by a factor of ~6.2x and ~18.9x for payload size 16 MB and 512 MB, respectively. This increase in the write latency negatively impacts the data collection time and leads to inefficiency in the overall training time.
To mitigate this, we use the sharding capability provided by Reverb to increase the throughput between actors, learner, and replay buffer services. Sharding balances the write load from the large number of actors across multiple replay buffer servers, instead of throttling a single replay buffer server, and also minimizes the average write latency for each replay buffer server (as fewer actors share the same server). This enables Menger to scale efficiently to thousands of actors across multiple Borg cells.
Case Study: Chip Placement We studied the benefits of Menger in the complex task of chip placement for a large netlist. Using 512 TPU cores, Menger achieves significant improvements in the training time (up to ~8.6x, reducing the training time from ~8.6 hours down to merely one hour in the fastest configuration) compared to a strong baseline. While Menger was optimized for TPUs, that the key factor for this performance gain is the architecture, and we would expect to see similar gains when tailored to use on GPUs.
We believe that Menger infrastructure and its promising results in the intricate task of chip placement demonstrate an innovative path forward to further shorten the chip design cycle and has the potential to not only enable further innovations in the chip design process, but other challenging real-world tasks as well.
Acknowledgments Most of the work was done by Amir Yazdanbakhsh, Junchao Chen, and Yu Zheng. We would like to also thank Robert Ormandi, Ebrahim Songhori, Shen Wang, TF-Agents team, Albin Cassirer, Aviral Kumar, James Laudon, John Wilkes, Joe Jiang, Milad Hashemi, Sat Chatterjee, Piotr Stanczyk, Sabela Ramos, Lasse Espeholt, Marcin Michalski, Sam Fishman, Ruoxin Sang, Azalia Mirhosseini, Anna Goldie, and Eric Johnson for their help and support.
1 A Menger cube is a three-dimensional fractal curve, and the inspiration for the name of this system, given that the proposed infrastructure can virtually scale ad infinitum. ↩
