A Medium article on Meta A.I. Speech to Speech Textless Spoken Language Translation.
https://medium.com/@sidsanc4998/translate-your-cats-meow-884d2bdd4587
Direct Punjabi to English Speech Translation using Discrete UnitsIJCI JOURNAL
Speech-to-speech translation is yet to reach the same level of coverage as text-to-text translation systems. The current speech technology is highly limited in its coverage of over 7000 languages spoken worldwide, leaving more than half of the population deprived of such technology and shared experiences. With voice-assisted technology (such as social robots and speech-to-text apps) and auditory content (such as podcasts and lectures) on the rise, ensuring that the technology is available for all is more important than ever. Speech translation can play a vital role in mitigating technological disparity and creating a more inclusive society. With a motive to contribute towards speech translation research for low-resource languages, our work presents a direct speech-to-speech translation model for one of the Indic languages called Punjabi to English. Additionally, we explore the performance of using a discrete representation of speech called discrete acoustic units as input to the Transformer-based translation model. The model, abbreviated as Unit-to-Unit Translation (U2UT), takes a sequence of discrete units of the source language (the language being translated from) and outputs a sequence of discrete units of the target language (the language being translated to). Our results show that the U2UT model performs better than the Speechto-Unit Translation (S2UT) model by a 3.69 BLEU score.
This document is the final report for a bachelor's degree project on developing an English language text-to-speech synthesis system. It summarizes the student's research and implementation of a diphone synthesis system that concatenates recorded speech segments. The student created methods for automatically extracting diphones from speech, constructing a diphone database, smoothing concatenations, and modifying prosody. Evaluation tests showed the system achieved over 76% intelligibility on various speech tests. The student discusses future work on the system, including improving naturalness.
Improvement in Quality of Speech associated with Braille codes - A Reviewinscit2006
J. Anurag, P. Nupur and Agrawal, S.S.
School of Information Technology, Guru Gobind Singh Indraprastha University, Delhi, India
Centre for Development of Advanced Computing, Noida, India
IRJET- Text to Speech Synthesis for Hindi Language using Festival FrameworkIRJET Journal
This document describes a text-to-speech synthesis system for the Hindi language developed using the Festival framework. The system takes Hindi text as input and outputs synthesized speech. It uses a syllable-based concatenative approach where Hindi words are segmented into syllables which are then matched to recorded audio files and concatenated to generate speech. Challenges in developing text-to-speech for Hindi include accurate pronunciation rules and producing natural prosody. The system aims to improve the naturalness of synthesized Hindi speech output.
The document discusses sequence to sequence models for speech recognition. It describes how traditional automatic speech recognition (ASR) works using acoustic, pronunciation, and language models. The document then introduces sequence to sequence models like Listen, Attend and Spell (LAS) which uses an encoder, attender, and decoder. LAS improves upon traditional ASR by integrating all models into a single neural network with attention and other optimizations like minimum word error rate training and scheduled sampling. Sequence to sequence models provide around 11% relative improvement in word error rate over traditional ASR systems.
EMPLOYING PIVOT LANGUAGE TECHNIQUE THROUGH STATISTICAL AND NEURAL MACHINE TRA...ijnlc
The quality of Neural Machine Translation (NMT) systems like Statistical Machine Translation (SMT) systems, heavily depends on the size of training data set, while for some pairs of languages, high-quality parallel data are poor resources. In order to respond to this low-resourced training data bottleneck reality, we employ the pivoting approach in both neural MT and statistical MT frameworks. During our experiments on the Persian-Spanish, taken as an under-resourced translation task, we discovered that, the aforementioned method, in both frameworks, significantly improves the translation quality in comparison to the standard direct translation approach.
IRJET- Tamil Speech to Indian Sign Language using CMUSphinx Language ModelsIRJET Journal
The document describes a proposed system to translate Tamil speech to Indian Sign Language (ISL) using speech recognition and natural language processing algorithms. It aims to help hearing-impaired people communicate independently. The system would use the CMU Sphinx speech recognition tool to convert spoken Tamil to text, then apply grammar rules and machine learning to translate the text to ISL displayed through video or animated avatars. The document reviews similar existing systems and research on speech recognition and sign language translation to inform the design and implementation of the proposed Tamil-ISL system.
Direct Punjabi to English Speech Translation using Discrete UnitsIJCI JOURNAL
Speech-to-speech translation is yet to reach the same level of coverage as text-to-text translation systems. The current speech technology is highly limited in its coverage of over 7000 languages spoken worldwide, leaving more than half of the population deprived of such technology and shared experiences. With voice-assisted technology (such as social robots and speech-to-text apps) and auditory content (such as podcasts and lectures) on the rise, ensuring that the technology is available for all is more important than ever. Speech translation can play a vital role in mitigating technological disparity and creating a more inclusive society. With a motive to contribute towards speech translation research for low-resource languages, our work presents a direct speech-to-speech translation model for one of the Indic languages called Punjabi to English. Additionally, we explore the performance of using a discrete representation of speech called discrete acoustic units as input to the Transformer-based translation model. The model, abbreviated as Unit-to-Unit Translation (U2UT), takes a sequence of discrete units of the source language (the language being translated from) and outputs a sequence of discrete units of the target language (the language being translated to). Our results show that the U2UT model performs better than the Speechto-Unit Translation (S2UT) model by a 3.69 BLEU score.
This document is the final report for a bachelor's degree project on developing an English language text-to-speech synthesis system. It summarizes the student's research and implementation of a diphone synthesis system that concatenates recorded speech segments. The student created methods for automatically extracting diphones from speech, constructing a diphone database, smoothing concatenations, and modifying prosody. Evaluation tests showed the system achieved over 76% intelligibility on various speech tests. The student discusses future work on the system, including improving naturalness.
Improvement in Quality of Speech associated with Braille codes - A Reviewinscit2006
J. Anurag, P. Nupur and Agrawal, S.S.
School of Information Technology, Guru Gobind Singh Indraprastha University, Delhi, India
Centre for Development of Advanced Computing, Noida, India
IRJET- Text to Speech Synthesis for Hindi Language using Festival FrameworkIRJET Journal
This document describes a text-to-speech synthesis system for the Hindi language developed using the Festival framework. The system takes Hindi text as input and outputs synthesized speech. It uses a syllable-based concatenative approach where Hindi words are segmented into syllables which are then matched to recorded audio files and concatenated to generate speech. Challenges in developing text-to-speech for Hindi include accurate pronunciation rules and producing natural prosody. The system aims to improve the naturalness of synthesized Hindi speech output.
The document discusses sequence to sequence models for speech recognition. It describes how traditional automatic speech recognition (ASR) works using acoustic, pronunciation, and language models. The document then introduces sequence to sequence models like Listen, Attend and Spell (LAS) which uses an encoder, attender, and decoder. LAS improves upon traditional ASR by integrating all models into a single neural network with attention and other optimizations like minimum word error rate training and scheduled sampling. Sequence to sequence models provide around 11% relative improvement in word error rate over traditional ASR systems.
EMPLOYING PIVOT LANGUAGE TECHNIQUE THROUGH STATISTICAL AND NEURAL MACHINE TRA...ijnlc
The quality of Neural Machine Translation (NMT) systems like Statistical Machine Translation (SMT) systems, heavily depends on the size of training data set, while for some pairs of languages, high-quality parallel data are poor resources. In order to respond to this low-resourced training data bottleneck reality, we employ the pivoting approach in both neural MT and statistical MT frameworks. During our experiments on the Persian-Spanish, taken as an under-resourced translation task, we discovered that, the aforementioned method, in both frameworks, significantly improves the translation quality in comparison to the standard direct translation approach.
IRJET- Tamil Speech to Indian Sign Language using CMUSphinx Language ModelsIRJET Journal
The document describes a proposed system to translate Tamil speech to Indian Sign Language (ISL) using speech recognition and natural language processing algorithms. It aims to help hearing-impaired people communicate independently. The system would use the CMU Sphinx speech recognition tool to convert spoken Tamil to text, then apply grammar rules and machine learning to translate the text to ISL displayed through video or animated avatars. The document reviews similar existing systems and research on speech recognition and sign language translation to inform the design and implementation of the proposed Tamil-ISL system.
This document presents a system for real time voice cloning using deep learning models. It discusses existing text-to-speech systems that are typically single speaker and proposes a system that can generate speech for different target speakers without observing them during training. The proposed system uses a speaker encoder to analyze a reference voice note, a synthesizer to generate text with the reference frequency, and a neural vocoder to produce a speech waveform. The document outlines the system architecture, requirements and concludes the goal of building a voice cloning system with data efficient natural speech generation for various speakers was achieved.
“Neural Machine Translation for low resource languages: Use case anglais - wolof“ by Sokhar Samb - Data scientist at @THEOLEX
Abstract : We will dive into the different steps of developing a Wolof-English machine translation using JoeyNMT using the benchmark from Masakhane NLP.
This presentation took place during a joint WiMLDS meetup between Paris & Dakar.
Approach of Syllable Based Unit Selection Text- To-Speech Synthesis System fo...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
MULTILINGUAL SPEECH IDENTIFICATION USING ARTIFICIAL NEURAL NETWORKijitcs
Speech technology is an emerging technology and automatic speech recognition has made advances in recent years. Many researches has been performed for many foreign and regional languages. But at present the multilingual speech processing technology has been attracting for research purpose. This paper tries to propose a methodology for developing a bilingual speech identification system for Assamese and English language based on artificial neural network.
Investigation of Text-to-Speech based Synthetic Parallel Data for Sequence-to...NU_I_TODALAB
APSIPA ASC 2021
Ding Ma, Wen-Chin Huang, Tomoki Toda: Investigation of text-to-speech-based synthetic parallel data for sequence-to-sequence non-parallel voice conversion, Dec. 2021
Toda Laboratory, Department of Intelligent Systems, Graduate School of Informatics, Nagoya University
SMATalk: Standard Malay Text to Speech Talk SystemCSCJournals
This document summarizes a research paper on SMaTTS, a rule-based text-to-speech synthesis system for Standard Malay. The system uses a sinusoidal synthesis method and some pre-recorded wave files to generate speech. It has two main phases: natural language processing (NLP) and digital signal processing (DSP). The NLP phase analyzes text and converts it to phonetic representations with prosody information. The DSP phase generates the speech waveform. The system was evaluated using diagnostic rhyme tests and mean opinion scores, and areas for improvement are discussed.
This document summarizes a seminar report on English to Assamese statistical machine translation using Moses. It includes sections on introduction to machine translation and statistical machine translation, implementation details of training Moses on an English-Assamese parallel corpus, results and evaluation using BLEU score, and proposed solutions to problems like handling out-of-vocabulary words through transliteration. The summary provides an overview of the topics and structure covered in the seminar report.
El modelo de traducción de voz de extremo a extremo de alta calidad se basa en una gran escala de datos de entrenamiento de voz a texto,
que suele ser escaso o incluso no está disponible para algunos pares de idiomas de bajos recursos. Para superar esto, nos
proponer un método de aumento de datos del lado del objetivo para la traducción del habla en idiomas de bajos recursos. En particular,
primero generamos paráfrasis del lado objetivo a gran escala basadas en un modelo de generación de paráfrasis
que incorpora varias características de traducción automática estadística (SMT) y el uso común
función de red neuronal recurrente (RNN). Luego, un modelo de filtrado que consiste en similitud semántica
y se propuso la co-ocurrencia de pares de palabras y habla para seleccionar la fuente con la puntuación más alta
pares de paráfrasis de los candidatos. Resultados experimentales en inglés, árabe, alemán, letón, estonio,
La generación de paráfrasis eslovena y sueca muestra que el método propuesto logra resultados significativos.
y mejoras consistentes sobre varios modelos de referencia sólidos en conjuntos de datos PPDB (http://paraphrase.
org/). Para introducir los resultados de la generación de paráfrasis en la traducción de voz de bajo recurso,
proponen dos estrategias: recombinación de pares audio-texto y entrenamiento de referencias múltiples. Experimental
Los resultados muestran que los modelos de traducción de voz entrenados en nuevos conjuntos de datos de audio y texto que combinan
los resultados de la generación de paráfrasis conducen a mejoras sustanciales sobre las líneas de base, especialmente en
lenguas de escasos recursos.
Modeling of Speech Synthesis of Standard Arabic Using an Expert Systemcsandit
This document describes an expert system for speech synthesis of Standard Arabic text. It involves two main stages: 1) creation of a sound database and 2) text-to-speech transformation. The transformation process involves phonetic orthographic transcription of the text and then generating voice signals corresponding to the transcribed phonetic sequence. The expert system uses a knowledge base containing sound data and rewriting rules. It transcribes text using graphemes as basic units and then concatenates sound units from the database to synthesize speech. Tests achieved a 96% success rate in pronouncing sentences correctly. Future work aims to improve prosody and develop fully automatic signal segmentation.
Searching for the Best Machine Translation CombinationMatīss
Matīss Rikters is researching hybrid machine translation methods. He used a count-based language model for candidate selection from full translations, combining translations of sentence chunks, and combining translations of linguistically motivated chunks. He also used a character-level neural language model for candidate selection. His methods achieved BLEU scores up to 19.51. Future work includes completing experiments on English-Estonian, winning the WMT17 news translation task for English-Latvian, performing chunking on the target side, and experimenting with other language models for candidate selection.
The document describes a voice activated text editing software project that uses speech recognition, speech synthesis, and machine learning based text summarization. The software allows users to create notes, import documents, and perform text functions using voice commands. It was created to reduce the time spent manually typing documents and to provide independent digital note-taking for visually impaired individuals. The system design and algorithms for extractive and abstractive text summarization are presented along with experimental results and comparisons to other systems.
MULTILINGUAL SPEECH TO TEXT CONVERSION USING HUGGING FACE FOR DEAF PEOPLEIRJET Journal
The document describes a system for multilingual speech-to-text conversion using Hugging Face that aims to assist deaf individuals. The system uses a Transformer-based encoder-decoder model trained on audio datasets. Feature extraction and tokenization are performed on the audio inputs. The model is fine-tuned using Hugging Face and evaluated based on word error rate. A web application allows users to record speech via microphone and see the transcription output. The implemented model achieved a 32.42% word error rate on a Hindi language dataset. The goal is to enable seamless communication for those with hearing impairments across multiple languages.
The document discusses developing a text-to-speech system that can generate voice outputs in various Hindi dialects given a transcript as input. It outlines developing modules for speech recognition and text analysis to determine the specified dialect. It explores both concatenative synthesis that joins recorded sounds and neural network models for audio synthesis. Future work includes collecting an annotated Hindi voice dataset to train an end-to-end recurrent neural network model for improved text-to-speech conversion across multiple Hindi dialects.
This document provides an overview of an example-based machine translation system that translates short Arabic sentences to English. It describes the key modules of the system, including preprocessing, matching, transfer, and recombination. The system uses a large parallel Arabic-English corpus extracted from United Nations documents to find translation examples and fragments. Evaluation results for the system are promising based on automatic metrics. Future work is needed to improve efficiency and perform full recombination of translated fragments into coherent English sentences.
Implementation of Marathi Language Speech Databases for Large Dictionaryiosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.
Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels
This document discusses speech synthesis technology. It begins with an introduction defining speech synthesis as the artificial production of human speech. It then discusses the history of speech synthesis, including early inventions and developments of speech synthesizers. It also covers the construction and various approaches to speech synthesis, such as concatenative synthesis and formant synthesis. The document concludes by discussing applications of speech synthesis and remaining challenges.
Real-time DirectTranslation System for Sinhala and Tamil Languages.Sheeyam Shellvacumar
Presented my research on "Real-time DirectTranslation System for Sinhala and Tamil Languages" at the FedCSIS 2015 Research Conference hosted by University of Lodz, Poland from 13 - 17th of September 2015.
IRJET- Speech to Speech Translation SystemIRJET Journal
1. The document describes a speech-to-speech translation system that aims to facilitate conversations between people speaking different languages.
2. It discusses the architecture of the proposed system, which includes modules for speech input, speech recognition, translation, grammar correction, text-to-speech synthesis, and speech output.
3. The document also reviews related work on speech recognition, translation, and text-to-speech systems. It outlines the implementation status of the different modules in the proposed system and possibilities for future improvement, such as supporting additional languages.
How to Interpret Trends in the Kalyan Rajdhani Mix Chart.pdfChart Kalyan
A Mix Chart displays historical data of numbers in a graphical or tabular form. The Kalyan Rajdhani Mix Chart specifically shows the results of a sequence of numbers over different periods.
This document presents a system for real time voice cloning using deep learning models. It discusses existing text-to-speech systems that are typically single speaker and proposes a system that can generate speech for different target speakers without observing them during training. The proposed system uses a speaker encoder to analyze a reference voice note, a synthesizer to generate text with the reference frequency, and a neural vocoder to produce a speech waveform. The document outlines the system architecture, requirements and concludes the goal of building a voice cloning system with data efficient natural speech generation for various speakers was achieved.
“Neural Machine Translation for low resource languages: Use case anglais - wolof“ by Sokhar Samb - Data scientist at @THEOLEX
Abstract : We will dive into the different steps of developing a Wolof-English machine translation using JoeyNMT using the benchmark from Masakhane NLP.
This presentation took place during a joint WiMLDS meetup between Paris & Dakar.
Approach of Syllable Based Unit Selection Text- To-Speech Synthesis System fo...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
MULTILINGUAL SPEECH IDENTIFICATION USING ARTIFICIAL NEURAL NETWORKijitcs
Speech technology is an emerging technology and automatic speech recognition has made advances in recent years. Many researches has been performed for many foreign and regional languages. But at present the multilingual speech processing technology has been attracting for research purpose. This paper tries to propose a methodology for developing a bilingual speech identification system for Assamese and English language based on artificial neural network.
Investigation of Text-to-Speech based Synthetic Parallel Data for Sequence-to...NU_I_TODALAB
APSIPA ASC 2021
Ding Ma, Wen-Chin Huang, Tomoki Toda: Investigation of text-to-speech-based synthetic parallel data for sequence-to-sequence non-parallel voice conversion, Dec. 2021
Toda Laboratory, Department of Intelligent Systems, Graduate School of Informatics, Nagoya University
SMATalk: Standard Malay Text to Speech Talk SystemCSCJournals
This document summarizes a research paper on SMaTTS, a rule-based text-to-speech synthesis system for Standard Malay. The system uses a sinusoidal synthesis method and some pre-recorded wave files to generate speech. It has two main phases: natural language processing (NLP) and digital signal processing (DSP). The NLP phase analyzes text and converts it to phonetic representations with prosody information. The DSP phase generates the speech waveform. The system was evaluated using diagnostic rhyme tests and mean opinion scores, and areas for improvement are discussed.
This document summarizes a seminar report on English to Assamese statistical machine translation using Moses. It includes sections on introduction to machine translation and statistical machine translation, implementation details of training Moses on an English-Assamese parallel corpus, results and evaluation using BLEU score, and proposed solutions to problems like handling out-of-vocabulary words through transliteration. The summary provides an overview of the topics and structure covered in the seminar report.
El modelo de traducción de voz de extremo a extremo de alta calidad se basa en una gran escala de datos de entrenamiento de voz a texto,
que suele ser escaso o incluso no está disponible para algunos pares de idiomas de bajos recursos. Para superar esto, nos
proponer un método de aumento de datos del lado del objetivo para la traducción del habla en idiomas de bajos recursos. En particular,
primero generamos paráfrasis del lado objetivo a gran escala basadas en un modelo de generación de paráfrasis
que incorpora varias características de traducción automática estadística (SMT) y el uso común
función de red neuronal recurrente (RNN). Luego, un modelo de filtrado que consiste en similitud semántica
y se propuso la co-ocurrencia de pares de palabras y habla para seleccionar la fuente con la puntuación más alta
pares de paráfrasis de los candidatos. Resultados experimentales en inglés, árabe, alemán, letón, estonio,
La generación de paráfrasis eslovena y sueca muestra que el método propuesto logra resultados significativos.
y mejoras consistentes sobre varios modelos de referencia sólidos en conjuntos de datos PPDB (http://paraphrase.
org/). Para introducir los resultados de la generación de paráfrasis en la traducción de voz de bajo recurso,
proponen dos estrategias: recombinación de pares audio-texto y entrenamiento de referencias múltiples. Experimental
Los resultados muestran que los modelos de traducción de voz entrenados en nuevos conjuntos de datos de audio y texto que combinan
los resultados de la generación de paráfrasis conducen a mejoras sustanciales sobre las líneas de base, especialmente en
lenguas de escasos recursos.
Modeling of Speech Synthesis of Standard Arabic Using an Expert Systemcsandit
This document describes an expert system for speech synthesis of Standard Arabic text. It involves two main stages: 1) creation of a sound database and 2) text-to-speech transformation. The transformation process involves phonetic orthographic transcription of the text and then generating voice signals corresponding to the transcribed phonetic sequence. The expert system uses a knowledge base containing sound data and rewriting rules. It transcribes text using graphemes as basic units and then concatenates sound units from the database to synthesize speech. Tests achieved a 96% success rate in pronouncing sentences correctly. Future work aims to improve prosody and develop fully automatic signal segmentation.
Searching for the Best Machine Translation CombinationMatīss
Matīss Rikters is researching hybrid machine translation methods. He used a count-based language model for candidate selection from full translations, combining translations of sentence chunks, and combining translations of linguistically motivated chunks. He also used a character-level neural language model for candidate selection. His methods achieved BLEU scores up to 19.51. Future work includes completing experiments on English-Estonian, winning the WMT17 news translation task for English-Latvian, performing chunking on the target side, and experimenting with other language models for candidate selection.
The document describes a voice activated text editing software project that uses speech recognition, speech synthesis, and machine learning based text summarization. The software allows users to create notes, import documents, and perform text functions using voice commands. It was created to reduce the time spent manually typing documents and to provide independent digital note-taking for visually impaired individuals. The system design and algorithms for extractive and abstractive text summarization are presented along with experimental results and comparisons to other systems.
MULTILINGUAL SPEECH TO TEXT CONVERSION USING HUGGING FACE FOR DEAF PEOPLEIRJET Journal
The document describes a system for multilingual speech-to-text conversion using Hugging Face that aims to assist deaf individuals. The system uses a Transformer-based encoder-decoder model trained on audio datasets. Feature extraction and tokenization are performed on the audio inputs. The model is fine-tuned using Hugging Face and evaluated based on word error rate. A web application allows users to record speech via microphone and see the transcription output. The implemented model achieved a 32.42% word error rate on a Hindi language dataset. The goal is to enable seamless communication for those with hearing impairments across multiple languages.
The document discusses developing a text-to-speech system that can generate voice outputs in various Hindi dialects given a transcript as input. It outlines developing modules for speech recognition and text analysis to determine the specified dialect. It explores both concatenative synthesis that joins recorded sounds and neural network models for audio synthesis. Future work includes collecting an annotated Hindi voice dataset to train an end-to-end recurrent neural network model for improved text-to-speech conversion across multiple Hindi dialects.
This document provides an overview of an example-based machine translation system that translates short Arabic sentences to English. It describes the key modules of the system, including preprocessing, matching, transfer, and recombination. The system uses a large parallel Arabic-English corpus extracted from United Nations documents to find translation examples and fragments. Evaluation results for the system are promising based on automatic metrics. Future work is needed to improve efficiency and perform full recombination of translated fragments into coherent English sentences.
Implementation of Marathi Language Speech Databases for Large Dictionaryiosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.
Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels
This document discusses speech synthesis technology. It begins with an introduction defining speech synthesis as the artificial production of human speech. It then discusses the history of speech synthesis, including early inventions and developments of speech synthesizers. It also covers the construction and various approaches to speech synthesis, such as concatenative synthesis and formant synthesis. The document concludes by discussing applications of speech synthesis and remaining challenges.
Real-time DirectTranslation System for Sinhala and Tamil Languages.Sheeyam Shellvacumar
Presented my research on "Real-time DirectTranslation System for Sinhala and Tamil Languages" at the FedCSIS 2015 Research Conference hosted by University of Lodz, Poland from 13 - 17th of September 2015.
IRJET- Speech to Speech Translation SystemIRJET Journal
1. The document describes a speech-to-speech translation system that aims to facilitate conversations between people speaking different languages.
2. It discusses the architecture of the proposed system, which includes modules for speech input, speech recognition, translation, grammar correction, text-to-speech synthesis, and speech output.
3. The document also reviews related work on speech recognition, translation, and text-to-speech systems. It outlines the implementation status of the different modules in the proposed system and possibilities for future improvement, such as supporting additional languages.
Similar to SiddhantSancheti_MediumShortStory.pptx (20)
How to Interpret Trends in the Kalyan Rajdhani Mix Chart.pdfChart Kalyan
A Mix Chart displays historical data of numbers in a graphical or tabular form. The Kalyan Rajdhani Mix Chart specifically shows the results of a sequence of numbers over different periods.
Nunit vs XUnit vs MSTest Differences Between These Unit Testing Frameworks.pdfflufftailshop
When it comes to unit testing in the .NET ecosystem, developers have a wide range of options available. Among the most popular choices are NUnit, XUnit, and MSTest. These unit testing frameworks provide essential tools and features to help ensure the quality and reliability of code. However, understanding the differences between these frameworks is crucial for selecting the most suitable one for your projects.
Unlock the Future of Search with MongoDB Atlas_ Vector Search Unleashed.pdfMalak Abu Hammad
Discover how MongoDB Atlas and vector search technology can revolutionize your application's search capabilities. This comprehensive presentation covers:
* What is Vector Search?
* Importance and benefits of vector search
* Practical use cases across various industries
* Step-by-step implementation guide
* Live demos with code snippets
* Enhancing LLM capabilities with vector search
* Best practices and optimization strategies
Perfect for developers, AI enthusiasts, and tech leaders. Learn how to leverage MongoDB Atlas to deliver highly relevant, context-aware search results, transforming your data retrieval process. Stay ahead in tech innovation and maximize the potential of your applications.
#MongoDB #VectorSearch #AI #SemanticSearch #TechInnovation #DataScience #LLM #MachineLearning #SearchTechnology
HCL Notes und Domino Lizenzkostenreduzierung in der Welt von DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-und-domino-lizenzkostenreduzierung-in-der-welt-von-dlau/
DLAU und die Lizenzen nach dem CCB- und CCX-Modell sind für viele in der HCL-Community seit letztem Jahr ein heißes Thema. Als Notes- oder Domino-Kunde haben Sie vielleicht mit unerwartet hohen Benutzerzahlen und Lizenzgebühren zu kämpfen. Sie fragen sich vielleicht, wie diese neue Art der Lizenzierung funktioniert und welchen Nutzen sie Ihnen bringt. Vor allem wollen Sie sicherlich Ihr Budget einhalten und Kosten sparen, wo immer möglich. Das verstehen wir und wir möchten Ihnen dabei helfen!
Wir erklären Ihnen, wie Sie häufige Konfigurationsprobleme lösen können, die dazu führen können, dass mehr Benutzer gezählt werden als nötig, und wie Sie überflüssige oder ungenutzte Konten identifizieren und entfernen können, um Geld zu sparen. Es gibt auch einige Ansätze, die zu unnötigen Ausgaben führen können, z. B. wenn ein Personendokument anstelle eines Mail-Ins für geteilte Mailboxen verwendet wird. Wir zeigen Ihnen solche Fälle und deren Lösungen. Und natürlich erklären wir Ihnen das neue Lizenzmodell.
Nehmen Sie an diesem Webinar teil, bei dem HCL-Ambassador Marc Thomas und Gastredner Franz Walder Ihnen diese neue Welt näherbringen. Es vermittelt Ihnen die Tools und das Know-how, um den Überblick zu bewahren. Sie werden in der Lage sein, Ihre Kosten durch eine optimierte Domino-Konfiguration zu reduzieren und auch in Zukunft gering zu halten.
Diese Themen werden behandelt
- Reduzierung der Lizenzkosten durch Auffinden und Beheben von Fehlkonfigurationen und überflüssigen Konten
- Wie funktionieren CCB- und CCX-Lizenzen wirklich?
- Verstehen des DLAU-Tools und wie man es am besten nutzt
- Tipps für häufige Problembereiche, wie z. B. Team-Postfächer, Funktions-/Testbenutzer usw.
- Praxisbeispiele und Best Practices zum sofortigen Umsetzen
Dive into the realm of operating systems (OS) with Pravash Chandra Das, a seasoned Digital Forensic Analyst, as your guide. 🚀 This comprehensive presentation illuminates the core concepts, types, and evolution of OS, essential for understanding modern computing landscapes.
Beginning with the foundational definition, Das clarifies the pivotal role of OS as system software orchestrating hardware resources, software applications, and user interactions. Through succinct descriptions, he delineates the diverse types of OS, from single-user, single-task environments like early MS-DOS iterations, to multi-user, multi-tasking systems exemplified by modern Linux distributions.
Crucial components like the kernel and shell are dissected, highlighting their indispensable functions in resource management and user interface interaction. Das elucidates how the kernel acts as the central nervous system, orchestrating process scheduling, memory allocation, and device management. Meanwhile, the shell serves as the gateway for user commands, bridging the gap between human input and machine execution. 💻
The narrative then shifts to a captivating exploration of prominent desktop OSs, Windows, macOS, and Linux. Windows, with its globally ubiquitous presence and user-friendly interface, emerges as a cornerstone in personal computing history. macOS, lauded for its sleek design and seamless integration with Apple's ecosystem, stands as a beacon of stability and creativity. Linux, an open-source marvel, offers unparalleled flexibility and security, revolutionizing the computing landscape. 🖥️
Moving to the realm of mobile devices, Das unravels the dominance of Android and iOS. Android's open-source ethos fosters a vibrant ecosystem of customization and innovation, while iOS boasts a seamless user experience and robust security infrastructure. Meanwhile, discontinued platforms like Symbian and Palm OS evoke nostalgia for their pioneering roles in the smartphone revolution.
The journey concludes with a reflection on the ever-evolving landscape of OS, underscored by the emergence of real-time operating systems (RTOS) and the persistent quest for innovation and efficiency. As technology continues to shape our world, understanding the foundations and evolution of operating systems remains paramount. Join Pravash Chandra Das on this illuminating journey through the heart of computing. 🌟
Your One-Stop Shop for Python Success: Top 10 US Python Development Providersakankshawande
Simplify your search for a reliable Python development partner! This list presents the top 10 trusted US providers offering comprehensive Python development services, ensuring your project's success from conception to completion.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Generating privacy-protected synthetic data using Secludy and MilvusZilliz
During this demo, the founders of Secludy will demonstrate how their system utilizes Milvus to store and manipulate embeddings for generating privacy-protected synthetic data. Their approach not only maintains the confidentiality of the original data but also enhances the utility and scalability of LLMs under privacy constraints. Attendees, including machine learning engineers, data scientists, and data managers, will witness first-hand how Secludy's integration with Milvus empowers organizations to harness the power of LLMs securely and efficiently.
Driving Business Innovation: Latest Generative AI Advancements & Success StorySafe Software
Are you ready to revolutionize how you handle data? Join us for a webinar where we’ll bring you up to speed with the latest advancements in Generative AI technology and discover how leveraging FME with tools from giants like Google Gemini, Amazon, and Microsoft OpenAI can supercharge your workflow efficiency.
During the hour, we’ll take you through:
Guest Speaker Segment with Hannah Barrington: Dive into the world of dynamic real estate marketing with Hannah, the Marketing Manager at Workspace Group. Hear firsthand how their team generates engaging descriptions for thousands of office units by integrating diverse data sources—from PDF floorplans to web pages—using FME transformers, like OpenAIVisionConnector and AnthropicVisionConnector. This use case will show you how GenAI can streamline content creation for marketing across the board.
Ollama Use Case: Learn how Scenario Specialist Dmitri Bagh has utilized Ollama within FME to input data, create custom models, and enhance security protocols. This segment will include demos to illustrate the full capabilities of FME in AI-driven processes.
Custom AI Models: Discover how to leverage FME to build personalized AI models using your data. Whether it’s populating a model with local data for added security or integrating public AI tools, find out how FME facilitates a versatile and secure approach to AI.
We’ll wrap up with a live Q&A session where you can engage with our experts on your specific use cases, and learn more about optimizing your data workflows with AI.
This webinar is ideal for professionals seeking to harness the power of AI within their data management systems while ensuring high levels of customization and security. Whether you're a novice or an expert, gain actionable insights and strategies to elevate your data processes. Join us to see how FME and AI can revolutionize how you work with data!
Skybuffer SAM4U tool for SAP license adoptionTatiana Kojar
Manage and optimize your license adoption and consumption with SAM4U, an SAP free customer software asset management tool.
SAM4U, an SAP complimentary software asset management tool for customers, delivers a detailed and well-structured overview of license inventory and usage with a user-friendly interface. We offer a hosted, cost-effective, and performance-optimized SAM4U setup in the Skybuffer Cloud environment. You retain ownership of the system and data, while we manage the ABAP 7.58 infrastructure, ensuring fixed Total Cost of Ownership (TCO) and exceptional services through the SAP Fiori interface.
Programming Foundation Models with DSPy - Meetup SlidesZilliz
Prompting language models is hard, while programming language models is easy. In this talk, I will discuss the state-of-the-art framework DSPy for programming foundation models with its powerful optimizers and runtime constraint system.
Fueling AI with Great Data with Airbyte WebinarZilliz
This talk will focus on how to collect data from a variety of sources, leveraging this data for RAG and other GenAI use cases, and finally charting your course to productionalization.
Presentation of the OECD Artificial Intelligence Review of Germany
SiddhantSancheti_MediumShortStory.pptx
1.
2. • Standard text-based translation systems are not enough in the
current world, where we have more than thousands of
languages. This is because the traditional systems have
drawbacks in creating speech-to-speech translation systems.
• It employs a cascading set of processes where the computing
costs and inference latency increase with each stage.
• This method cannot be used to translate into every spoken
language because more than 40% of the languages in the
world lack text writing systems..
A Direct Speech-to-Speech
Translate (S2ST)
3. Meta Version
of Direct S2ST
Advancing S2ST with discrete
units
• Enables faster inference and supports translation
between unwritten languages.
• It does not rely on text generation as an intermediate
step
• Trained using actual, publicly available audio data
instead of synthetic audio for numerous language pairs.
• The researchers used discretized speech units instead
of spectrograms, which they derived by clustering self-
supervised speech representations.
4. Meta’s Grip
over translate
Much Faster
and Better
The S2ST system performs
better than earlier direct S2ST
systems
Trained on real
data
It is first direct S2ST system to
be trained on real S2ST data
for many language pairings
Use of
Pretraining
It makes use of pretraining with
unlabeled speech data.
5. Mark with a better
solution
• The researchers employed self-supervised discrete units as targets (speech-to-unit
translation, or S2UT) for training the direct S2ST system to facilitate direct speech-to-
speech translation with discrete units (audio samples).
• They suggest a transformer-based sequence-to-sequence paradigm with an integrated
voice encoder and discrete unit decoder
6. Models and Improvements
S2ST model with discrete units.
A transformer-based S2UT model with a
speech encoder and a discrete unit
decoder
Flowchart and Finetuning process
Speech encoder and decoder
Two-pass decoding mechanism
The first-pass decoder generates text in
a related language (Mandarin), and the
second-pass decoder creates units.
7. Illustration of the textless S2ST model
• The left side is the speech-to-unit translation (S2UT) model with an auxiliary task while the right part is the unit-
based HiFi-GAN vocoder for unit-to-speech conversion.
8. Experiment Results
Average 3.2 BLEU gain when training the S2ST model on
the VoxPopuli S2ST dataset, compared to a baseline trained
on un-normalized speech target. Theye also incorporated
automatically mined S2ST data.
S2ST model that predicts using discrete units results
outperforms
6.6-12.1
BLEU
gain
additional
2.0 BLEU
gain
9. Experiment Data:
• Their study uses the Fisher Spanish-English speech translation corpus, which comprises 139K sentences (about
170 hours) transcribed in both Spanish and English from Spanish-speaking telephone conversations.
• For modeling target speech in English, Spanish or French, they train a single mHuBERT model with 100k subset of
VoxPopuli unlabeled speech, which contains 4.5k hrs of data from three languages for En, Es, and Fr.
• They employed VoxPopuli ASR dataset and convert text transcriptions to reference units for training the speech
normalizer. TTS data for HiFi-GAN vocoder along VAD to remove the silence at both ends of the audio
https://github.com/pytorch/fairseq/blob/ main/examples/speech_to_speech/docs/textless_ s2st_real_data.md
https://huggingface.co/facebook/tts_ transformer-en-ljspeech, Es: https://huggingface. co/facebook/tts_transformer-es-css10
10. Future of Translation
Simultaneous
translation
Large collection of S2ST
d e v e l o p e d t h r o u g h o u r
innovative NLP toolkit called
LASER.
SpeechMatrix
Building high-quality S2ST
models without any human
annotations.
Unsupervised Learning
Break down language barriers
in both the physical world and
the metaverse
Handshake between realms
11. References
“Enhanced Direct Speech-to-
Speech Translation Using
Self-supervised Pre-training
and Data Augmentation”
https://arxiv.org/abs/2204.02967
“Direct Speech-to-Speech
Translation With Discrete
Units”
https://arxiv.org/abs/2107.05604
“Textless Speech-to-
Speech Translation on
Real Data”
https://arxiv.org/abs/2112.08352
“Speech-to-speech
translation between
untranscribed unknown
languages”
https://arxiv.org/abs/1910.00795