Archive for the ‘Machine Learning’ Category

Machine learning can reduce risk of antibiotic resistance in chickens and humans as well – Innovation Origins

Why we write about this topic:

The rapid increase in poultry production has resulted in extensive and indiscriminate use of antibiotics. This has led to a worrying increase in cases of antimicrobial resistance which could potentially spread to humans.

Scientists have used machine learning to find new ways to identify and pinpoint disease in poultry farms, which will help to reduce the need for antibiotic treatment, lowering the risk of antibiotic resistance transferring to human populations, says the University of Nottingham in a press release.

The study, published inSpringer Nature, was led by Dr. Tania Dottorini at the University of Nottingham. The research is part of the FARMWATCH project, a 1.5m partnership between the University and the China National Center for Food Safety Risk Assessment.

With antibiotic resistance now one of the most threatening issues worldwide, effective and rapid diagnostics of bacterial infection in chicken farming can reduce the need for antibiotics, which will reduce the risk of epidemics and antibiotic resistance.

In this project, researchers in Nottingham collected samples from the animals, humans and environment in a Chinese farm and connected slaughterhouse. This complex big data has now been analysed for new diagnostic that will predict and detect bacterial infection, insurgence of antimicrobial resistance (AMR), and transfer to humans. This data will then allow early intervention and treatment, reducing spread and the need for antibiotics.

Smart software discovers new types of antibiotics Innovation Origins

A research group at Leiden University is using artificial intelligence (AI) to find new types of antibiotics. The software could replace the old cultivation method.

The study produced three key findings. Firstly, several similar clinically relevant antimicrobial resistance genes (ARGs) and associated mobile genetic elements (antibiotic resistance genes able to move within genomes and between bacteria), were found in both human and broiler chicken samples.

Secondly, by developing a machine learning-powered approach the team found the existence of a core chicken gut resistome that is correlated with the AMR circulating in the farms. Finally, using sensing technology and machine learning, the team uncovered that the AMR-related core resistome are themselves associated with various external factors such as temperature and humidity.

Dr Dottorini said: The food production industry represents a major consumer of antibiotics, but the AMR risks within these environments are still not fully understood. It is therefore critical to set out studies and improved methods optimised to these environments where animals and humans may be in close contact. Precision farming, cost-effective DNA sequencing and the increased adoption of machine learning technologies offer the opportunity to develop methods giving a better understanding and quantification of AMR risks in farming environments.

Supercomputers joined the fight against antibiotic resistance Innovation Origins

The development of antibiotics is one of the most significant breakthroughs in medicine. However, pathogens develop resistance mechanisms that thwart the effectiveness of antibiotics.

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Machine learning can reduce risk of antibiotic resistance in chickens and humans as well - Innovation Origins

Locus granted patent for machine learning models to ‘accurately predict traffic time’ for last-mile deliveries – Robotics and Automation News

Locus, a logistics software technology company (not to be confused with Locus Robotics), has been awarded a patent for Machine Learning Models for Predicting Time in Traffic, the second patent awarded to the company in 2022.

Predictability of last-mile deliveries is a logistical challenge that continues to affect businesses across various sectors such as e-commerce, 3PLs, and so on.

Locus new patent will enable enterprises to achieve better precision in their last-mile deliveries by factoring in traffic patterns, which have historically been considered too dynamic to map.

The patent covers unique technology that analyzes the historical data of traffic and predicts the travel time between origin and destination locations.

It also factors in sub-variables such as day of the week and time of day to provide hyper-accurate estimated travel times for logistics providers to get ahead.

Geet Garg, founder and CTO, Locus, says: To constantly add value to our growing customer base across 30+ geographies, we at Locus, are pushing our engineering boundaries each day.

The latest patent highlights our relentless effort to bring out new levels of innovation in our product suite, enabling enterprises across industries to drive high-precision logistics operations in any geography, reduce costs and achieve business success.

To foster a culture rooted in innovation, Locus rolled out a company-wide program in 2018 that empowers all employees to apply for patents and includes regular training sessions, end-to-end support for IPR filings, and much more.

Through this initiative, Locus has secured four patents in the last four years, with additional filings already in the pipeline that collectively will continue to bring advancements in the companys unique product suite to serve its customers better.

Nishith Rastogi, founder and CEO of Locus, says: Innovation lives within our DNA and we believe that every employee should have an equal opportunity to contribute to the next best idea.

Thus, with the goal to cultivate an environment of entrepreneurship and creativity, we introduced this policy to empower and support our team in their commitment to solving challenges of the last-mile logistics.

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Locus granted patent for machine learning models to 'accurately predict traffic time' for last-mile deliveries - Robotics and Automation News

Deep Learning Could Bring the Concert Experience Home – IEEE Spectrum

Now that recorded sound has become ubiquitous, we hardly think about it. From our smartphones, smart speakers, TVs, radios, disc players, and car sound systems, its an enduring and enjoyable presence in our lives. In 2017, a survey by the polling firm Nielsen suggested that some 90 percent of the U.S. population listens to music regularly and that, on average, they do so 32 hours per week.

Behind this free-flowing pleasure are enormous industries applying technology to the long-standing goal of reproducing sound with the greatest possible realism. From Edisons phonograph and the horn speakers of the 1880s, successive generations of engineers in pursuit of this ideal invented and exploited countless technologies: triode vacuum tubes, dynamic loudspeakers, magnetic phonograph cartridges, solid-state amplifier circuits in scores of different topologies, electrostatic speakers, optical discs, stereo, and surround sound. And over the past five decades, digital technologies, like audio compression and streaming, have transformed the music industry.

And yet even now, after 150 years of development, the sound we hear from even a high-end audio system falls far short of what we hear when we are physically present at a live music performance. At such an event, we are in a natural sound field and can readily perceive that the sounds of different instruments come from different locations, even when the sound field is criss-crossed with mixed sound from multiple instruments. Theres a reason why people pay considerable sums to hear live music: It is more enjoyable, exciting, and can generate a bigger emotional impact.

Today, researchers, companies, and entrepreneurs, including ourselves, are closing in at last on recorded audio that truly re-creates a natural sound field. The group includes big companies, such as Apple and Sony, as well as smaller firms, such as Creative. Netflix recently disclosed a partnership with Sennheiser under which the network has begun using a new system, Ambeo 2-Channel Spatial Audio, to heighten the sonic realism of such TV shows as Stranger Things and The Witcher.

There are now at least half a dozen different approaches to producing highly realistic audio. We use the term soundstage to distinguish our work from other audio formats, such as the ones referred to as spatial audio or immersive audio. These can represent sound with more spatial effect than ordinary stereo, but they do not typically include the detailed sound-source location cues that are needed to reproduce a truly convincing sound field.

We believe that soundstage is the future of music recording and reproduction. But before such a sweeping revolution can occur, it will be necessary to overcome an enormous obstacle: that of conveniently and inexpensively converting the countless hours of existing recordings, regardless of whether theyre mono, stereo, or multichannel surround sound (5.1, 7.1, and so on). No one knows exactly how many songs have been recorded, but according to the entertainment-metadata concern Gracenote, more than 200 million recorded songs are available now on planet Earth. Given that the average duration of a song is about 3 minutes, this is the equivalent of about 1,100 years of music.

That is a lot of music. Any attempt to popularize a new audio format, no matter how promising, is doomed to fail unless it includes technology that makes it possible for us to listen to all this existing audio with the same ease and convenience with which we now enjoy stereo musicin our homes, at the beach, on a train, or in a car.

We have developed such a technology. Our system, which we call 3D Soundstage, permits music playback in soundstage on smartphones, ordinary or smart speakers, headphones, earphones, laptops, TVs, soundbars, and in vehicles. Not only can it convert mono and stereo recordings to soundstage, it also allows a listener with no special training to reconfigure a sound field according to their own preference, using a graphical user interface. For example, a listener can assign the locations of each instrument and vocal sound source and adjust the volume of eachchanging the relative volume of, say, vocals in comparison with the instrumental accompaniment. The system does this by leveraging artificial intelligence (AI), virtual reality, and digital signal processing (more on that shortly).

To re-create convincingly the sound coming from, say, a string quartet in two small speakers, such as the ones available in a pair of headphones, requires a great deal of technical finesse. To understand how this is done, lets start with the way we perceive sound.

When sound travels to your ears, unique characteristics of your headits physical shape, the shape of your outer and inner ears, even the shape of your nasal cavitieschange the audio spectrum of the original sound. Also, there is a very slight difference in the arrival time from a sound source to your two ears. From this spectral change and the time difference, your brain perceives the location of the sound source. The spectral changes and time difference can be modeled mathematically as head-related transfer functions (HRTFs). For each point in three-dimensional space around your head, there is a pair of HRTFs, one for your left ear and the other for the right.

So, given a piece of audio, we can process that audio using a pair of HRTFs, one for the right ear, and one for the left. To re-create the original experience, we would need to take into account the location of the sound sources relative to the microphones that recorded them. If we then played that processed audio back, for example through a pair of headphones, the listener would hear the audio with the original cues, and perceive that the sound is coming from the directions from which it was originally recorded.

If we dont have the original location information, we can simply assign locations for the individual sound sources and get essentially the same experience. The listener is unlikely to notice minor shifts in performer placementindeed, they might prefer their own configuration.

Even now, after 150 years of development, the sound we hear from even a high-end audio system falls far short of what we hear when we are physically present at a live music performance.

There are many commercial apps that use HRTFs to create spatial sound for listeners using headphones and earphones. One example is Apples Spatialize Stereo. This technology applies HRTFs to playback audio so you can perceive a spatial sound effecta deeper sound field that is more realistic than ordinary stereo. Apple also offers a head-tracker version that uses sensors on the iPhone and AirPods to track the relative direction between your head, as indicated by the AirPods in your ears, and your iPhone. It then applies the HRTFs associated with the direction of your iPhone to generate spatial sounds, so you perceive that the sound is coming from your iPhone. This isnt what we would call soundstage audio, because instrument sounds are still mixed together. You cant perceive that, for example, the violin player is to the left of the viola player.

Apple does, however, have a product that attempts to provide soundstage audio: Apple Spatial Audio. It is a significant improvement over ordinary stereo, but it still has a couple of difficulties, in our view. One, it incorporates Dolby Atmos, a surround-sound technology developed by Dolby Laboratories. Spatial Audio applies a set of HRTFs to create spatial audio for headphones and earphones. However, the use of Dolby Atmos means that all existing stereophonic music would have to be remastered for this technology. Remastering the millions of songs already recorded in mono and stereo would be basically impossible. Another problem with Spatial Audio is that it can only support headphones or earphones, not speakers, so it has no benefit for people who tend to listen to music in their homes and cars.

So how does our system achieve realistic soundstage audio? We start by using machine-learning software to separate the audio into multiple isolated tracks, each representing one instrument or singer or one group of instruments or singers. This separation process is called upmixing. A producer or even a listener with no special training can then recombine the multiple tracks to re-create and personalize a desired sound field.

Consider a song featuring a quartet consisting of guitar, bass, drums, and vocals. The listener can decide where to locate the performers and can adjust the volume of each, according to his or her personal preference. Using a touch screen, the listener can virtually arrange the sound-source locations and the listeners position in the sound field, to achieve a pleasing configuration. The graphical user interface displays a shape representing the stage, upon which are overlaid icons indicating the sound sourcesvocals, drums, bass, guitars, and so on. There is a head icon at the center, indicating the listeners position. The listener can touch and drag the head icon around to change the sound field according to their own preference.

Moving the head icon closer to the drums makes the sound of the drums more prominent. If the listener moves the head icon onto an icon representing an instrument or a singer, the listener will hear that performer as a solo. The point is that by allowing the listener to reconfigure the sound field, 3D Soundstage adds new dimensions (if youll pardon the pun) to the enjoyment of music.

The converted soundstage audio can be in two channels, if it is meant to be heard through headphones or an ordinary left- and right-channel system. Or it can be multichannel, if it is destined for playback on a multiple-speaker system. In this latter case, a soundstage audio field can be created by two, four, or more speakers. The number of distinct sound sources in the re-created sound field can even be greater than the number of speakers.

This multichannel approach should not be confused with ordinary 5.1 and 7.1 surround sound. These typically have five or seven separate channels and a speaker for each, plus a subwoofer (the .1). The multiple loudspeakers create a sound field that is more immersive than a standard two-speaker stereo setup, but they still fall short of the realism possible with a true soundstage recording. When played through such a multichannel setup, our 3D Soundstage recordings bypass the 5.1, 7.1, or any other special audio formats, including multitrack audio-compression standards.

A word about these standards. In order to better handle the data for improved surround-sound and immersive-audio applications, new standards have been developed recently. These include the MPEG-H 3D audio standard for immersive spatial audio with Spatial Audio Object Coding (SAOC). These new standards succeed various multichannel audio formats and their corresponding coding algorithms, such as Dolby Digital AC-3 and DTS, which were developed decades ago.

While developing the new standards, the experts had to take into account many different requirements and desired features. People want to interact with the music, for example by altering the relative volumes of different instrument groups. They want to stream different kinds of multimedia, over different kinds of networks, and through different speaker configurations. SAOC was designed with these features in mind, allowing audio files to be efficiently stored and transported, while preserving the possibility for a listener to adjust the mix based on their personal taste.

To do so, however, it depends on a variety of standardized coding techniques. To create the files, SAOC uses an encoder. The inputs to the encoder are data files containing sound tracks; each track is a file representing one or more instruments. The encoder essentially compresses the data files, using standardized techniques. During playback, a decoder in your audio system decodes the files, which are then converted back to the multichannel analog sound signals by digital-to-analog converters.

Our 3D Soundstage technology bypasses this. We use mono or stereo or multichannel audio data files as input. We separate those files or data streams into multiple tracks of isolated sound sources, and then convert those tracks to two-channel or multichannel output, based on the listeners preferred configurations, to drive headphones or multiple loudspeakers. We use AI technology to avoid multitrack rerecording, encoding, and decoding.

In fact, one of the biggest technical challenges we faced in creating the 3D Soundstage system was writing that machine-learning software that separates (or upmixes) a conventional mono, stereo, or multichannel recording into multiple isolated tracks in real time. The software runs on a neural network. We developed this approach for music separation in 2012 and described it in patents that were awarded in 2022 and 2015 (the U.S. patent numbers are 11,240,621 B2 and 9,131,305 B2).

The listener can decide where to locate the performers and can adjust the volume of each, according to his or her personal preference.

A typical session has two components: training and upmixing. In the training session, a large collection of mixed songs, along with their isolated instrument and vocal tracks, are used as the input and target output, respectively, for the neural network. The training uses machine learning to optimize the neural-network parameters so that the output of the neural networkthe collection of individual tracks of isolated instrument and vocal datamatches the target output.

A neural network is very loosely modeled on the brain. It has an input layer of nodes, which represent biological neurons, and then many intermediate layers, called hidden layers. Finally, after the hidden layers there is an output layer, where the final results emerge. In our system, the data fed to the input nodes is the data of a mixed audio track. As this data proceeds through layers of hidden nodes, each node performs computations that produce a sum of weighted values. Then a nonlinear mathematical operation is performed on this sum. This calculation determines whether and how the audio data from that node is passed on to the nodes in the next layer.

There are dozens of these layers. As the audio data goes from layer to layer, the individual instruments are gradually separated from one another. At the end, in the output layer, each separated audio track is output on a node in the output layer.

Thats the idea, anyway. While the neural network is being trained, the output may be off the mark. It might not be an isolated instrumental trackit might contain audio elements of two instruments, for example. In that case, the individual weights in the weighting scheme used to determine how the data passes from hidden node to hidden node are tweaked and the training is run again. This iterative training and tweaking goes on until the output matches, more or less perfectly, the target output.

As with any training data set for machine learning, the greater the number of available training samples, the more effective the training will ultimately be. In our case, we needed tens of thousands of songs and their separated instrumental tracks for training; thus, the total training music data sets were in the thousands of hours.

After the neural network is trained, given a song with mixed sounds as input, the system outputs the multiple separated tracks by running them through the neural network using the system established during training.

After separating a recording into its component tracks, the next step is to remix them into a soundstage recording. This is accomplished by a soundstage signal processor. This soundstage processor performs a complex computational function to generate the output signals that drive the speakers and produce the soundstage audio. The inputs to the generator include the isolated tracks, the physical locations of the speakers, and the desired locations of the listener and sound sources in the re-created sound field. The outputs of the soundstage processor are multitrack signals, one for each channel, to drive the multiple speakers.

The sound field can be in a physical space, if it is generated by speakers, or in a virtual space, if it is generated by headphones or earphones. The function performed within the soundstage processor is based on computational acoustics and psychoacoustics, and it takes into account sound-wave propagation and interference in the desired sound field and the HRTFs for the listener and the desired sound field.

For example, if the listener is going to use earphones, the generator selects a set of HRTFs based on the configuration of desired sound-source locations, then uses the selected HRTFs to filter the isolated sound-source tracks. Finally, the soundstage processor combines all the HRTF outputs to generate the left and right tracks for earphones. If the music is going to be played back on speakers, at least two are needed, but the more speakers, the better the sound field. The number of sound sources in the re-created sound field can be more or less than the number of speakers.

We released our first soundstage app, for the iPhone, in 2020. It lets listeners configure, listen to, and save soundstage music in real timethe processing causes no discernible time delay. The app, called 3D Musica, converts stereo music from a listeners personal music library, the cloud, or even streaming music to soundstage in real time. (For karaoke, the app can remove vocals, or output any isolated instrument.)

Earlier this year, we opened a Web portal, 3dsoundstage.com, that provides all the features of the 3D Musica app in the cloud plus an application programming interface (API) making the features available to streaming music providers and even to users of any popular Web browser. Anyone can now listen to music in soundstage audio on essentially any device.

When sound travels to your ears, unique characteristics of your headits physical shape, the shape of your outer and inner ears, even the shape of your nasal cavitieschange the audio spectrum of the original sound.

We also developed separate versions of the 3D Soundstage software for vehicles and home audio systems and devices to re-create a 3D sound field using two, four, or more speakers. Beyond music playback, we have high hopes for this technology in videoconferencing. Many of us have had the fatiguing experience of attending videoconferences in which we had trouble hearing other participants clearly or being confused about who was speaking. With soundstage, the audio can be configured so that each person is heard coming from a distinct location in a virtual room. Or the location can simply be assigned depending on the persons position in the grid typical of Zoom and other videoconferencing applications. For some, at least, videoconferencing will be less fatiguing and speech will be more intelligible.

Just as audio moved from mono to stereo, and from stereo to surround and spatial audio, it is now starting to move to soundstage. In those earlier eras, audiophiles evaluated a sound system by its fidelity, based on such parameters as bandwidth, harmonic distortion, data resolution, response time, lossless or lossy data compression, and other signal-related factors. Now, soundstage can be added as another dimension to sound fidelityand, we dare say, the most fundamental one. To human ears, the impact of soundstage, with its spatial cues and gripping immediacy, is much more significant than incremental improvements in fidelity. This extraordinary feature offers capabilities previously beyond the experience of even the most deep-pocketed audiophiles.

Technology has fueled previous revolutions in the audio industry, and it is now launching another one. Artificial intelligence, virtual reality, and digital signal processing are tapping in to psychoacoustics to give audio enthusiasts capabilities theyve never had. At the same time, these technologies are giving recording companies and artists new tools that will breathe new life into old recordings and open up new avenues for creativity. At last, the century-old goal of convincingly re-creating the sounds of the concert hall has been achieved.

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Deep Learning Could Bring the Concert Experience Home - IEEE Spectrum

Winners of the Inaugural WLA Prize Announced RMB 10 Million for Each Prize – GlobeNewswire

Shanghai, Sept. 30, 2022 (GLOBE NEWSWIRE) -- The 2022 WLA Prize Laureates are:

--"For fundamental contributions to the foundations of machine learning and its application."

About the Laureate

Prof. Michael I. Jordan has been a world-leading researcher in the field of statistical machine learning for nearly four decades. His contributions to the interface between computer science and statistics include the variational approach to statistical inference and learning, inference methods based on graphical models and Bayesian non-parametrics, and characterizations of trade-offs between statistical risk and computational complexity.

He has also worked at the interface between optimization and machine learning, where he is well known for his development of continuous-time models of gradient-based optimization and sampling, and his work on distributed systems for optimization. He has built bridges between machine learning and control theory, contributing to the theory of reinforcement learning, learning-based model predictive control, and optimality principles for human motor control.

He has also led the way in bringing microeconomic concepts into contact with machine learning, developing learning methods that incentivize learners to share data, showing how contract theory can be employed for statistical inference, and contributing to the study of learning-based matching markets.He has also pursued numerous high-impact applications of machine learning in domains such as single-molecule imaging, protein modeling, genetic admixture modeling, and natural language processing.

Prof. Jordan's contributions to computer science are also evident in education. He has mentored over 80 PhD students and over 60 postdoctoral researchers, an influential cohort who are now professors at the world's leading academic institutions and leaders in the industry.

--"For key discoveries elucidating the mechanism and selectivity of protein transport between the cytoplasm and nucleus."

About the Laureate

Dr. Dirk Grlich was born in Halle/Saale in Germany and studied biochemistry at Martin Luther University in Halle. For his Ph.D., he joined Tom Rapoport's lab in Berlin, where he identified the heterotrimeric Sec61 complex as a receptor for translating ribosomes and protein-conducting channels of the endoplasmic reticulum (ER). He also succeeded in reconstituting a fully functional "translocon" from purified components and demonstrated its capacity to transport secretory proteins across the ER membrane and to integrate type I and type II membrane proteins into the lipid bilayer.

In 1993, Dr. Grlich joined Ron. Laskey's lab at the University of Cambridge, where he discovered the first importins as mediators of protein import into the cell nucleus.

In 1996, he became an independent group leader and later a professor of molecular biology at the ZMBH (University of Heidelberg). During this time, he developed the RanGTP-gradient model to explain the directionality and energetics of nuclear transport. His group first described chaperone functions of importins and was instrumental in discovering and characterizing exportins that mediate export from the cell nucleus.

Dr. Grlich is now a director at the Max Planck Institute for Multidisciplinary Sciences in Gttingen, focusing on the question of how nuclear pore complexes function as highly efficient transport machines. His team discovered that intrinsically disordered FG domains assemble into a condensed (selective) FG phase that serves as a highly selective permeability barrier of extreme transport capacity. His group also develops nanobodies as cell biological tools and, more recently, also as therapeutics for treating diseases such as Covid-19, malaria, bacterial infections, sepsis, and autoimmune conditions.

Roger KORNBERG, Chairman of the WorldLaureatesAssociation and 2006 Nobel Laureate in Chemistry, said that The WLA Prize advocates for original basic science and encourages scientific researchers to be better committed to the common well-being of human beings. It is believed that the WLA Prize, established in China through multi-lateral efforts, will become one of the world's globally influential awards.

Michael LEVITT, Vice Chairman of the World Laureates Association and 2013 Nobel Laureate in Chemistry announced the 2022 WLA Prize laureates.YANG Wei, Member of the Chinese Academy of Sciences, Foreign Member of the National Academy of Engineering (USA), and Fellow and Treasurer of the World Academy of Sciences (TWAS) also attended the press conference.

Global technological development has brought revolutionary and far-reaching impact since the start of this century, and the pace of development is only accelerating, said Neil SHEN, StewardofSequoiaCapitalFoundingandManagingPartnerofSequoiaChina, which is the exclusive sponsor of the WLA Prize. Sequoia China is committed to supporting cutting-edge technology and fundamental research and encouraging those who push forward the frontiers of science.

WU Xiangdong , Executive Director of the World Laureates Association and Chairman of the WLA Prize Management Committee announced that the Award Ceremony of the inaugural WLA Prize will be solemnly held at the Opening Ceremony of the 5th WLA Forum in early November 2022; both of the laureates are on schedule to Shanghai to accept their awards.

About the WLA Prize

The World Laureates Association Prize (WLA Prize) is an international science prize established in Shanghai, in 2021, initiated by World Laureates Association (WLA), managed by the WLA Foundation, and exclusively funded by Sequoia China.

The WLA Prize aims to recognize and support eminent researchers and technologists worldwide for their contributions to science. It is intended to support global science and technology advancement, address the challenges to humanity, and promote society's long-term progress.

Each year, the WLA Prize is awarded in two categories: the "WLA Prize in Computer Science or Mathematics" and the "WLA Prize in Life Science or Medicine."

The total award for each Prize, which may be divided among up to four laureates, is RMB 10 million.

More info about the WLA Prize:

http://www.wlaprize.org

About Us

World Laureates Association

The World Laureates Association (WLA) is a non-governmental and non-profit international organization. It is one of the world's highest profile organizations of laureates with three missions: advocacy for basic science, promotion of international cooperation, and support for young scientists. Upholding the vision of "Science and Technology for the Common Destiny of Mankind," the WLA is committed to enhancing the academic exchange of ideas and research among scientists and scholars in China and the world.

Follow Us for updates on Facebook and Twitter@wlaforum

WLA Foundation

The WLA Foundationsupports all WLA activities and is funded entirely by donations from private sources.

Sequoia China

Sequoia China helps daring founders build legendary companies. In partnering with Sequoia, companies benefit from its unmatched community and the lessons Sequoia has learned over the years. As "The Entrepreneurs Behind The Entrepreneurs", Sequoia China focuses on three sectors: tech, consumer and healthcare. Over the past 17 years, Sequoia China has had the privilege of working with more than 900 companies. It is committed to advancing innovation in science and technology. It actively supports scientists and entrepreneurs, promotes technological innovation, and nurtures leading tech-enabled enterprises, with a concern for corporate social responsibility. As the exclusive sponsor of the World Laureates Association (WLA) Prize, Sequoia China hopes that its support will encourage the pursuit of innovative scientific developments and drive growth across the world.

Link:
Winners of the Inaugural WLA Prize Announced RMB 10 Million for Each Prize - GlobeNewswire

Yash Prabhu celebrating victory after publication of research paper – Knight Crier

From taking home gold in the Delaware Valley Science Fair competition his sophomore year to having one of his most time-consuming and difficult research papers rejected at a regional competition Yash Prabhu turns his failure into a success story.

Yash Prabhu starts the school year strong with one of his research papers being published at a conference called IEEE. IEEE is an organization dedicated to assisting humanity through enhancing technology. The organization sponsors over 2,000 events yearly and organizes top-tier conferences with published papers being recognized by academia and industries. Prabhu recently had a paper published called A CNN Based Automated Stuttering Identification System, where he uses machine learning to detect different types of stuttering in audio segments.

Stuttering affects a lot of people around the world and its a surprisingly ignored issue. Some people who stutter receive speech therapy but a lot of people dont. Stuttering can damage the quality of life for people. It makes it harder to speak and harder to interact with people. You could get bullied, teachers could get frustrated speaking to you [and] it gets worse when youre older because you have to give meetings and presentations, explained senior Yash Prabhu.

The goal for his research paper revolves around his wish to make low-cost automation available to people worldwide, mainly in developing countries such as India where resources are scarce.

I am trying to design a model that can classify stuttering. By classifying stuttering it can help keep data. In an area where theres a lack of speech pathologists like in India, this model can have a big impact by keeping diagnostics on stuttering and also helping speech pathologists do a better and faster job so they can treat more people, Prabhu said.

The model Prabhu used is called a machine learning model. Machine learning creates functions to help make predictions.

I found a data set provided by Apple called SEP-28k and this data set [consists] of many examples of stuttering. I chose 5 different speech disfluencies and I trained my model to detect the speech disfluencies, Prabhu added.

Prabhu took the initiative to explore a data set that had not been extensively researched leading to hardships.

I emailed professors, I called doctors, I asked for access to data sets. Nothing really [worked] out because it was hard to find someone who could help me get access to data. A lot of doctors and professors are busy, they dont have time to talk to you, Prabhu said disappointed.

Nonetheless, his chances of success improved when Dr. Naeem Seliya, a professor at the University of Wisconsin Eau Claire, agreed to assist him.

I emailed him my credentials and asked him if I could work with him. He is actually a stutterer and he told me about a device he uses for stuttering. From there this idea was born in my mind. I can use my machine learning skills to have an impact in this area. At that point, I had no clue how to do it. I had absolutely no clue, I just knew I wanted to. For months I did research. I tried finding different ways to do it, but I failed a lot of times, Prabhu explained.

The difficulties he encountered while working with audio data resulted in poor models that produced faulty results.

When I submitted this to the local science fair it didnt go through which was a disappointment. I thought this was the most difficult project I have ever done to this day, Prabhu said disheartened.

He reflected back on his Sophomore year, where he designed a Covid-19 screener using machine learning and hardware. The project was inspired by the pandemic and received gold at the Delaware Valley Science Fair. However, his recent paper was rejected by the local science fair.

Being rejected by the local science fair was a big bruisebut I think it was a blessing in disguise because I got to make this whole paper. This paper was born from the science fair project failing. I made my models better, I trained them with more data, [and] I ended up submitting them to the conference, Prabhu added.

In the future, Prabhu plans to attend a four year university and continue doing research while immersing himself in robotics, engineering, and programming.

Continued here:
Yash Prabhu celebrating victory after publication of research paper - Knight Crier