摘要翻译:
在自动语音识别(ASR)中,用户所说的话取决于她所处的特定环境。通常,这个上下文表示为一组单词n-图。在这项工作中,我们提出了一个新的,全神经,端到端(E2E)ASR系统利用这样的上下文。我们的方法,我们重新命名为上下文监听、旁听和拼写(CLAS),联合优化了ASR组件以及上下文N-图的嵌入。在推理过程中,CLAS系统可以呈现上下文短语,其中可能包含在训练过程中看不到的词汇表外(OOV)术语。我们将我们所提出的系统与一种更传统的上下文化方法进行了比较,该方法在波束搜索过程中对内部训练的LAS和上下文n-gram模型进行了浅融合。在许多任务中,我们发现提出的CLAS系统比基线方法的相对WER多达68%,表明联合优化优于单独训练的组件。索引术语:语音识别,序列到序列模型,听,参加和拼写,LAS,注意,嵌入式语音识别。
---
英文标题:
《Deep context: end-to-end contextual speech recognition》
---
作者:
Golan Pundak, Tara N. Sainath, Rohit Prabhavalkar, Anjuli Kannan, Ding
Zhao
---
最新提交年份:
2018
---
分类信息:
一级分类:Electrical Engineering and Systems Science 电气工程与系统科学
二级分类:Audio and Speech Processing 音频和语音处理
分类描述:Theory and methods for processing signals representing audio, speech, and language, and their applications. This includes analysis, synthesis, enhancement, transformation, classification and interpretation of such signals as well as the design, development, and evaluation of associated signal processing systems. Machine learning and pattern analysis applied to any of the above areas is also welcome. Specific topics of interest include: auditory modeling and hearing aids; acoustic beamforming and source localization; classification of acoustic scenes; speaker separation; active noise control and echo cancellation; enhancement; de-reverberation; bioacoustics; music signals analysis, synthesis and modification; music information retrieval; audio for multimedia and joint audio-video processing; spoken and written language modeling, segmentation, tagging, parsing, understanding, and translation; text mining; speech production, perception, and psychoacoustics; speech analysis, synthesis, and perceptual modeling and coding; robust speech recognition; speaker recognition and characterization; deep learning, online learning, and graphical models applied to speech, audio, and language signals; and implementation aspects ranging from system architecture to fast algorithms.
处理代表音频、语音和语言的信号的理论和方法及其应用。这包括分析、合成、增强、转换、分类和解释这些信号,以及相关信号处理系统的设计、开发和评估。机器学习和模式分析应用于上述任何领域也是受欢迎的。感兴趣的具体主题包括:听觉建模和助听器;声波束形成与声源定位;声场景分类;说话人分离;有源噪声控制和回声消除;增强;去混响;生物声学;音乐信号的分析、合成与修饰;音乐信息检索;多媒体音频和联合音视频处理;口语和书面语建模、切分、标注、句法分析、理解和翻译;文本挖掘;言语产生、感知和心理声学;语音分析、合成、感知建模和编码;鲁棒语音识别;说话人识别与特征描述;应用于语音、音频和语言信号的
深度学习、在线学习和图形模型;以及从系统架构到快速算法的实现方面。
--
一级分类:Computer Science 计算机科学
二级分类:Machine Learning
机器学习
分类描述:Papers on all aspects of machine learning research (supervised, unsupervised, reinforcement learning, bandit problems, and so on) including also robustness, explanation, fairness, and methodology. cs.LG is also an appropriate primary category for applications of machine learning methods.
关于机器学习研究的所有方面的论文(有监督的,无监督的,强化学习,强盗问题,等等),包括健壮性,解释性,公平性和方法论。对于机器学习方法的应用,CS.LG也是一个合适的主要类别。
--
一级分类:Computer Science 计算机科学
二级分类:Sound 声音
分类描述:Covers all aspects of computing with sound, and sound as an information channel. Includes models of sound, analysis and synthesis, audio user interfaces, sonification of data, computer music, and sound signal processing. Includes ACM Subject Class H.5.5, and intersects with H.1.2, H.5.1, H.5.2, I.2.7, I.5.4, I.6.3, J.5, K.4.2.
涵盖了声音计算的各个方面,以及声音作为一种信息通道。包括声音模型、分析和合成、音频用户界面、数据的可听化、计算机音乐和声音信号处理。包括ACM学科类H.5.5,并与H.1.2、H.5.1、H.5.2、I.2.7、I.5.4、I.6.3、J.5、K.4.2交叉。
--
一级分类:Statistics 统计学
二级分类:Machine Learning 机器学习
分类描述:Covers machine learning papers (supervised, unsupervised, semi-supervised learning, graphical models, reinforcement learning, bandits, high dimensional inference, etc.) with a statistical or theoretical grounding
覆盖机器学习论文(监督,无监督,半监督学习,图形模型,强化学习,强盗,高维推理等)与统计或理论基础
--
---
英文摘要:
In automatic speech recognition (ASR) what a user says depends on the particular context she is in. Typically, this context is represented as a set of word n-grams. In this work, we present a novel, all-neural, end-to-end (E2E) ASR sys- tem that utilizes such context. Our approach, which we re- fer to as Contextual Listen, Attend and Spell (CLAS) jointly- optimizes the ASR components along with embeddings of the context n-grams. During inference, the CLAS system can be presented with context phrases which might contain out-of- vocabulary (OOV) terms not seen during training. We com- pare our proposed system to a more traditional contextualiza- tion approach, which performs shallow-fusion between inde- pendently trained LAS and contextual n-gram models during beam search. Across a number of tasks, we find that the pro- posed CLAS system outperforms the baseline method by as much as 68% relative WER, indicating the advantage of joint optimization over individually trained components. Index Terms: speech recognition, sequence-to-sequence models, listen attend and spell, LAS, attention, embedded speech recognition.
---
PDF链接:
https://arxiv.org/pdf/1808.0248