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Register a new userSpeechformer: Reducing Information Loss in Direct Speech Translation
الكلام: الحد من فقدان المعلومات في ترجمة الكلام المباشر
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اكتسبت النماذج المستندة إلى المحولات شعبية متزايدة تحقق الأداء الحديث في العديد من مجالات البحث بما في ذلك ترجمة الكلام. ومع ذلك، فإن التعقيد التربيعي للمحول فيما يتعلق بديل تسلسل الإدخال يمنع اعتماده كما هو الحال مع إشارات صوتية، والتي تمثل عادة بالتسلسلات الطويلة. اللجوء الحلول الحالية إلى ضغط أساسي أساسي على أساس أخذ العينات الثابتة لميزات الصوت الخام. لذلك، فإن المعلومات اللغوية المفيدة المحتملة غير قابلة للوصول إلى طبقات أعلى مستوى في الهندسة المعمارية. لحل هذه المشكلة، نقترح الكلمة، وهي عبارة عن بنية، بفضل انخفاض استخدام الذاكرة في طبقات الاهتمام، وتجنب ضغط المخلفات الأولية والمعلومات المجامعة فقط على مستوى أعلى وفقا لمعايير لغوية أكثر استنارة. تظهر التجارب على ثلاث أزواج اللغات (en → de / es / nl) فعالية حلنا، مع مكاسب تصل إلى 0.8 بلو على Corpus القياسية يجب أن تكون ميديا وعلى ما يصل إلى 4.0 بلو في سيناريو منخفض الموارد.
Transformer-based models have gained increasing popularity achieving state-of-the-art performance in many research fields including speech translation. However, Transformer's quadratic complexity with respect to the input sequence length prevents its adoption as is with audio signals, which are typically represented by long sequences. Current solutions resort to an initial sub-optimal compression based on a fixed sampling of raw audio features. Therefore, potentially useful linguistic information is not accessible to higher-level layers in the architecture. To solve this issue, we propose Speechformer, an architecture that, thanks to reduced memory usage in the attention layers, avoids the initial lossy compression and aggregates information only at a higher level according to more informed linguistic criteria. Experiments on three language pairs (en→de/es/nl) show the efficacy of our solution, with gains of up to 0.8 BLEU on the standard MuST-C corpus and of up to 4.0 BLEU in a low resource scenario.
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This paper describes the offline and simultaneous speech translation systems developed at AppTek for IWSLT 2021. Our offline ST submission includes the direct end-to-end system and the so-called posterior tight integrated model, which is akin to the
cascade system but is trained in an end-to-end fashion, where all the cascaded modules are end-to-end models themselves. For simultaneous ST, we combine hybrid automatic speech recognition with a machine translation approach whose translation policy decisions are learned from statistical word alignments. Compared to last year, we improve general quality and provide a wider range of quality/latency trade-offs, both due to a data augmentation method making the MT model robust to varying chunk sizes. Finally, we present a method for ASR output segmentation into sentences that introduces a minimal additional delay.
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