Papers
Topics
Authors
Recent
Gemini 2.5 Flash
Gemini 2.5 Flash
80 tokens/sec
GPT-4o
59 tokens/sec
Gemini 2.5 Pro Pro
43 tokens/sec
o3 Pro
7 tokens/sec
GPT-4.1 Pro
50 tokens/sec
DeepSeek R1 via Azure Pro
28 tokens/sec
2000 character limit reached

Towards end-to-end spoken language understanding (1802.08395v1)

Published 23 Feb 2018 in cs.CL

Abstract: Spoken language understanding system is traditionally designed as a pipeline of a number of components. First, the audio signal is processed by an automatic speech recognizer for transcription or n-best hypotheses. With the recognition results, a natural language understanding system classifies the text to structured data as domain, intent and slots for down-streaming consumers, such as dialog system, hands-free applications. These components are usually developed and optimized independently. In this paper, we present our study on an end-to-end learning system for spoken language understanding. With this unified approach, we can infer the semantic meaning directly from audio features without the intermediate text representation. This study showed that the trained model can achieve reasonable good result and demonstrated that the model can capture the semantic attention directly from the audio features.

Towards End-to-End Spoken Language Understanding

The paper addresses the challenge of enhancing spoken language understanding (SLU) systems through an end-to-end learning approach. Typically, SLU systems are structured as a pipeline incorporating automatic speech recognition (ASR) followed by natural language understanding (NLU). This tradition can lead to inefficiencies as each component is independently optimized, often resulting in error propagation. The authors propose a novel architecture combining these components to infer semantic intent directly from audio features, aiming to bypass errors inherent in text representation conversion.

Study Motivation and Traditional Approaches

Conventional SLU systems involve a serial processing approach where the audio input is initially transcribed by an ASR system, whose output is then analyzed by an NLU component for domain, intent, and slot extraction tasks. The major drawback lies in the separate optimization of ASR (minimizing word error rate) and NLU (trained on clean text), which can lead to a performance dip in noisy environments as transcription errors propagate. Moreover, human cognitive models for speech processing focus directly on concept extraction from speech, supporting the rationale for a direct audio-to-meaning framework.

Proposed End-to-End SLU Model

The end-to-end model leverages recurrent neural networks, specifically using a multi-layer bidirectional gated recurrent unit (GRU) network to process audio inputs represented as log-Mel filterbank features. The architecture circumvents intermediate text representation, aiming for direct intent classification. A notable feature is the inclusion of sub-sampling within GRUs to address long input sequences and mitigate computational overhead, crucial for real-time applications.

Empirical Evaluation

The performance of the end-to-end model was assessed on an industrial-scale dataset aligning with the structure of ATIS corpus. For domain classification, the proposed model closely matched transcript-based NLU models, indicating its capacity to capture high-level semantic cues. Intent classification posed a more intricate challenge, where the end-to-end model achieved competitive performance with a significantly compact architecture, underscoring its efficiency and potential scalability.

Noise robustness was specifically evaluated, showcasing a significant drop in traditional models' performance, whereas the end-to-end system maintained a relative robustness, highlighting its advantage in error-prone real-world scenarios.

Key Findings and Implications

  1. Compact Architecture: The end-to-end model demonstrates a substantial reduction in architectural complexity with only 0.4M parameters compared to 15.5M in conventional setups, making it viable for memory-constrained applications.
  2. Performance Trade-offs: While achieving slightly lower accuracy compared to text-input methods, the proposed approach shows promise in conditions where ASR might falter due to noise.
  3. Future Directions: Research on including slot filling in an end-to-end context is suggested, potentially through attention mechanisms, which could enhance SLU task integration, equipping the model to handle simultaneous word and slot predictions.

Conclusion

This research opens discussions on methodological shifts towards integrated SLU models capable of handling semantic understanding directly from audio inputs. While further enhancements are necessary to perfect performance parity with traditional methods, the presented work lays foundational strategies towards minimizing error propagation and improving system robustness in real-world applications. Future work might leverage deeper architectures and innovations in audio feature representation to realize comprehensive SLU systems.

User Edit Pencil Streamline Icon: https://streamlinehq.com
Authors (6)
  1. Dmitriy Serdyuk (20 papers)
  2. Yongqiang Wang (92 papers)
  3. Christian Fuegen (36 papers)
  4. Anuj Kumar (58 papers)
  5. Baiyang Liu (5 papers)
  6. Yoshua Bengio (601 papers)
Citations (228)