Accessibility Automatic Speech Recognition Computer-assisted transcription Confidence measures Docencia en Red Education language model adaptation Language Modeling Language Technologies Length modelling Log-linear models Machine Translation Massive Adaptation Modelat de la longitud Models basats en seqüències de paraules Models log-lineals Multilingualism Neural Machine Translation Opencast Matterhorn Polimedia Sliding window Speaker adaptation Speech Recognition Speech Translation Statistical machine translation streaming text-to-speech transcripciones video lecture repositories Video Lectures
2023 |
Iranzo Sánchez, Javier Streaming Neural Speech Translation PhD Thesis Universitat Politècnica de València, 2023, (Advisors: Alfons Juan Ciscar and Jorge Civera Saiz). Abstract | Links | BibTeX | Tags: Speech Translation, streaming speech translation @phdthesis{Sánchez2023, title = {Streaming Neural Speech Translation}, author = {Iranzo Sánchez, Javier}, doi = {10.4995/Thesis/10251/199170}, year = {2023}, date = {2023-09-29}, school = {Universitat Politècnica de València}, abstract = {Thanks to significant advances in Deep Learning, Speech Translation (ST) has become a mature field that enables the use of ST technology in production-ready solutions. Due to the ever-increasing hours of audio-visual content produced each year, as well as higher awareness of the importance of media accessibility, ST is poised to become a key element for the production of entertainment and educational media. Although significant advances have been made in ST, most research has focused on the offline scenario, where the entire input audio is available. In contrast, online ST remains an under-researched topic. A special case of online ST, streaming ST, translates an unbounded input stream in a real-time fashion under strict latency constraints. This is a much more realistic problem that needs to be solved in order to apply ST to a variety of real-life tasks. The focus of this thesis is on researching and developing key techniques necessary for a successful streaming ST solution. First, in order to enable ST system development and evaluation, a new multilingual ST dataset is collected, which significantly expands the amount of hours available for ST. Then, a streaming-ready segmenter component is developed to segment the intermediate transcriptions of our proposed cascade solution, which consists in an Automatic Speech Recognition (ASR) system that transcribes the audio, followed by a Machine Translation (MT) system that translates the intermediate transcriptions into the desired language. Research has shown that segmentation quality plays a significant role in downstream MT performance, so the development of an effective streaming segmenter is a critical step in the streaming ST process. This segmenter is then integrated and the components of the cascade are jointly optimized to achieve an appropriate quality-latency trade-off. Streaming ST has much more strict latency constraints than standard online ST, as the desired latency level must be maintained during the whole translation process. Therefore, it is crucial to be able to accurately measure this latency, but the standard online ST metrics are not well suited for this task. As a consequence, new evaluation methods are proposed for streaming ST evaluation, which ensure realistic, yet interpretable results. Lastly, a novel method is presented for improving translation quality through the use of contextual information. Whereas standard online ST systems translate audios in isolation, there is a wealth of contextual information available for improving streaming ST systems. Our approach introduces the concept of streaming history by storing the most recent information of the translation process, which is then used by the model in order to improve translation quality.}, note = {Advisors: Alfons Juan Ciscar and Jorge Civera Saiz}, keywords = {Speech Translation, streaming speech translation}, pubstate = {published}, tppubtype = {phdthesis} } Thanks to significant advances in Deep Learning, Speech Translation (ST) has become a mature field that enables the use of ST technology in production-ready solutions. Due to the ever-increasing hours of audio-visual content produced each year, as well as higher awareness of the importance of media accessibility, ST is poised to become a key element for the production of entertainment and educational media. Although significant advances have been made in ST, most research has focused on the offline scenario, where the entire input audio is available. In contrast, online ST remains an under-researched topic. A special case of online ST, streaming ST, translates an unbounded input stream in a real-time fashion under strict latency constraints. This is a much more realistic problem that needs to be solved in order to apply ST to a variety of real-life tasks. The focus of this thesis is on researching and developing key techniques necessary for a successful streaming ST solution. First, in order to enable ST system development and evaluation, a new multilingual ST dataset is collected, which significantly expands the amount of hours available for ST. Then, a streaming-ready segmenter component is developed to segment the intermediate transcriptions of our proposed cascade solution, which consists in an Automatic Speech Recognition (ASR) system that transcribes the audio, followed by a Machine Translation (MT) system that translates the intermediate transcriptions into the desired language. Research has shown that segmentation quality plays a significant role in downstream MT performance, so the development of an effective streaming segmenter is a critical step in the streaming ST process. This segmenter is then integrated and the components of the cascade are jointly optimized to achieve an appropriate quality-latency trade-off. Streaming ST has much more strict latency constraints than standard online ST, as the desired latency level must be maintained during the whole translation process. Therefore, it is crucial to be able to accurately measure this latency, but the standard online ST metrics are not well suited for this task. As a consequence, new evaluation methods are proposed for streaming ST evaluation, which ensure realistic, yet interpretable results. Lastly, a novel method is presented for improving translation quality through the use of contextual information. Whereas standard online ST systems translate audios in isolation, there is a wealth of contextual information available for improving streaming ST systems. Our approach introduces the concept of streaming history by storing the most recent information of the translation process, which is then used by the model in order to improve translation quality. |