IP Library Granted Patent US 12,354,597
Granted Patent B2
US 12,354,597 · App. 17/822,673 · Granted Jul 8, 2025

Disfluency detection models for natural conversational voice systems

Inventors: Shuo-yiin Chang (Sunnyvale, CA); Bo Li (Fremont, CA); Tara N. Sainath (Jersey City, NJ); Trevor Strohman (Mountain View, CA); Chao Zhang (Mountain View, CA)
Assignee: Google LLC
G10L15/063G10L15/083
View Patent ↗
Loading inventors, assignments & file history…
Monitor This Case
Get email alerts when status or documents change.
Order Certified Copies
Most orders are placed with the USPTO same day — all within 24 business hours.
Order via The Patent Place →
Pre-filled with this patent's details
Quick Facts
Patent No.
US 12,354,597
App. No.
17/822,673
Granted
Jul 8, 2025
Kind
B2
Abstract

A method includes receiving a sequence of acoustic frames characterizing one or more utterances. At each of a plurality of output steps, the method also includes generating, by an encoder network of a speech recognition model, a higher order feature representation for a corresponding acoustic frame of the sequence of acoustic frames, generating, by a prediction network of the speech recognition model, a hidden representation for a corresponding sequence of non-blank symbols output by a final softmax layer of the speech recognition model, and generating, by a first joint network of the speech recognition model that receives the higher order feature representation generated by the encoder network and the dense representation generated by the prediction network, a probability distribution that the corresponding time step corresponds to a pause and an end of speech.

Claims (78)

1. A computer-implemented method executed on data processing hardware that causes the data processing hardware to perform operations comprising:

receiving a sequence of acoustic frames characterizing one or more utterances; and

at each of a plurality of time steps:

generating, by an encoder network of a speech recognition model, a higher order feature representation for a corresponding acoustic frame in the sequence of acoustic frames;

outputting, by a final softmax layer of the speech recognition model, a sequence of non-blank output symbols;

generating, by a prediction network of the speech recognition model, a hidden representation for the sequence of non-blank output symbols;

generating, by a first joint network of the speech recognition model that receives the higher order feature representation generated by the encoder network and the hidden representation generated by the prediction network, a first probability distribution that the corresponding time step corresponds to a disfluency and an end of speech; and

generating, by a second joint network of the speech recognition model different from the first joint network that receives the higher order feature representation generated by the encoder network and the hidden representation generated by the prediction network, a second probability distribution over possible speech recognition hypotheses, wherein the final softmax layer outputs, based on the second probability distribution, a next non-blank output symbol.

2. The computer-implemented method of claim 1 , wherein the operations further comprise:

determining that the probability that the corresponding time step corresponds to the end of speech satisfies an end of speech threshold; and

in response to determining that the probability that the corresponding time step corresponds to the end of speech satisfies the end of speech threshold, triggering a microphone closing event.

3. The computer-implemented method of claim 1 , wherein the operations further comprise:

determining that the probability that the corresponding time step corresponds to the disfluency satisfies a disfluency threshold; and

emitting a disfluency token at the corresponding time step based on the determining that the probability of the corresponding time step corresponds to the disfluency satisfies the disfluency threshold.

4. The computer-implemented method of claim 1 , wherein the speech recognition model is trained by a two-stage training process, the two-stage training process comprising:

a first stage that trains the encoder network, the prediction network, and the second joint network on a speech recognition task; and

a second stage that initializes and fine-tunes the first joint network to learn how to predict pause and end of speech locations in utterances.

5. The computer-implemented method of claim 4 , wherein parameters of the encoder network, the prediction network, and the second joint network are frozen during the second stage of the two-stage training process.

6. The computer-implemented method of claim 4 , wherein the two-stage training process trains the speech recognition model on a plurality of transcribed training utterances having labels indicating pause and end of speech locations.

7. The computer-implemented method of claim 1 , wherein the encoder network comprises a stack of self-attention blocks.

8. The computer-implemented method of claim 7 , wherein the stack of self-attention blocks comprises a stack of conformer blocks or a stack of transformer blocks.

9. The computer-implemented method of claim 1 , wherein generating the hidden representation for the sequence of non-blank output symbols comprises:

for each non-blank output symbol in the sequence of non-blank output symbols received as input at the corresponding time step:

generating, by the prediction network, using a shared embedding matrix, an embedding of the non-blank output symbol;

assigning, by the prediction network, a respective position vector to the non-blank output symbol; and

weighting, by the prediction network, the embedding proportional to a similarity between the embedding and the respective position vector; and

generating, as output from the prediction network, a single embedding vector at the corresponding time step, the single embedding vector based on a weighted average of the weighted embeddings, the single embedding vector comprising the hidden representation.

10. The computer-implemented method of claim 9 , wherein the prediction network comprises a multi-headed attention mechanism, the multi-headed attention mechanism sharing the shared embedding matrix across each head of the multi-headed attention mechanism.

11. A system comprising:

data processing hardware; and

memory hardware in communication with the data processing hardware and storing instructions that, when executed on the data processing hardware, causes the data processing hardware to perform operations comprising:

receiving a sequence of acoustic frames characterizing one or more utterances; and

at each of a plurality of output steps:

generating, by an encoder network of a speech recognition model, a higher order feature representation for a corresponding acoustic frame in the sequence of acoustic frames;

outputting, by a final softmax layer of the speech recognition model, a sequence of non-blank output symbols;

generating, by a prediction network of the speech recognition model, a hidden representation for the sequence of non-blank output symbols;

generating, by a first joint network of the speech recognition model that receives the higher order feature representation generated by the encoder network and the hidden representation generated by the prediction network, a first probability distribution that the corresponding time step corresponds to a disfluency and an end of speech; and

generating, by a second joint network of the speech recognition model different from the first joint network that receives the higher order feature representation generated by the encoder network and the hidden representation generated by the prediction network, a second probability distribution over possible speech recognition hypotheses, wherein the final softmax layer outputs, based on the second probability distribution, a next non-blank output symbol.

12. The system of claim 11 , wherein the operations further comprise:

determining that the probability that the corresponding time step corresponds to the end of speech satisfies an end of speech threshold; and

in response to determining that the probability that the corresponding time step corresponds to the end of speech satisfies the end of speech threshold, triggering a microphone closing event.

13. The system of claim 11 , wherein the operations further comprise:

determining that the probability that the corresponding time step corresponds to the disfluency satisfies a disfluency threshold; and

emitting a disfluency token at the corresponding time step based on the determining that the probability of the corresponding time step corresponds to the disfluency satisfies the disfluency threshold.

14. The system of claim 11 , wherein the speech recognition model is trained by a two-stage training process, the two-stage training process comprising:

a first stage that trains the encoder network, the prediction network, and the second joint network on a speech recognition task; and

a second stage that initializes and fine-tunes the first joint network to learn how to predict pause and end of speech locations in utterances.

15. The system of claim 14 , wherein parameters of the encoder network, the prediction network, and the second joint network are frozen during the second stage of the two-stage training process.

16. The system of claim 14 , wherein the two-stage training process trains the speech recognition model on a plurality of transcribed training utterances having labels indicating pause and end of speech locations.

17. The system of claim 11 , wherein the encoder network comprises a stack of self-attention blocks.

18. The system of claim 17 , wherein the stack of self-attention blocks comprises a stack of conformer blocks or a stack of transformer blocks.

19. The system of claim 11 , wherein generating the hidden representation for the sequence of non-blank output symbols comprises:

for each non-blank output symbol in the sequence of non-blank output symbols received as input at the corresponding time step:

generating, by the prediction network, using a shared embedding matrix, an embedding of the non-blank output symbol;

assigning, by the prediction network, a respective position vector to the non-blank output symbol; and

weighting, by the prediction network, the embedding proportional to a similarity between the embedding and the respective position vector; and

generating, as output from the prediction network, a single embedding vector at the corresponding time step, the single embedding vector based on a weighted average of the weighted embeddings, the single embedding vector comprising the hidden representation.

20. The system of claim 19 , wherein the prediction network comprises a multi-headed attention mechanism, the multi-headed attention mechanism sharing the shared embedding matrix across each head of the multi-headed attention mechanism.

21. A system comprising:

data processing hardware; and

memory hardware in communication with the data processing hardware and storing instructions that, when executed on the data processing hardware, causes the data processing hardware to perform operations for executing a natural conversation automated speech recognition (ASR) model comprising:

an encoder network configured to:

receive, as input, a sequence of acoustic frames characterizing one or more utterances; and

generate, at each of a plurality of time steps, a higher order feature representation for a corresponding acoustic frame in the sequence of acoustic frames;

a final softmax layer configured to output a sequence of non-blank output symbols;

a prediction network configured to:

receive, as input, the sequence of non-blank output symbols output by the final softmax layer; and

generate, at each of the plurality of time steps, a hidden representation;

a first joint network configured to:

receive, as input, the hidden representation generated by the prediction network at each of the plurality of time steps and the higher order feature representation generated by the encoder network at each of the plurality of time steps; and

generate, at each of the plurality of time steps, a first probability distribution that the corresponding time step corresponds to a disfluency and an end of speech; and

a second joint network different from the first joint network and configured to:

receive the higher order feature representation generated by the encoder network and the hidden representation generated by the prediction network; and

generate, at each of the plurality of time steps, a second probability distribution over possible speech recognition hypotheses, wherein the final softmax layer outputs, based on the second probability distribution, a next non-blank output symbol.

22. The system of claim 21 , wherein the ASR model triggers a microphone closing event based on the probability that the corresponding time step corresponds to end of speech satisfying a threshold.

23. The system of claim 21 , wherein:

the encoder network, the prediction network, and the second joint network are trained on a speech recognition task during a first training stage; and

after the first training stage, the first joint network is initialized and fine-tuned to learn how to predict pause and end of speech locations in utterances while parameters of the encoder network, the prediction network, and the second joint network are frozen.

Assignments (2)
CORRECTIVE ASSIGNMENT TO CORRECT THE FIRST INVENTOR'S FIRST NAME PREVIOUSLY RECORDED AT REEL: 60921 FRAME: 911. ASSIGNOR(S) HEREBY CONFIRMS THE ASSIGNMENT . Recorded Jan 3, 2025
From: CHANG, SHUO-YIIN; LI, BO; SAINATH, TARA N.; STROHMAN, TREVOR; ZHANG, CHAO
To: GOOGLE LLC
Reel/Frame 069850/0815 →
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Aug 29, 2022
From: CHANG, SHO-YIIN; LI, BO; SAINATH, TARA N.; STROHMAN, TREVOR; ZHANG, CHAO
To: GOOGLE LLC
Reel/Frame 060921/0911 →
Continuity (2)
Provisional Application 63262185 · Oct 6, 2021
Related Publication 20230107450A1 · Apr 6, 2023
References Cited (19)
US 10923111B1 · Fan · 2021 [cited by examiner]
US 12002451B1 · Liu · 2024 [cited by examiner]
US 20170270919A1 · Parthasarathi · 2017 [cited by examiner]
US 20180350395A1 · Simko · 2018 [cited by examiner]
US 20190325898A1 · O'Hart Kinney et al. · 2019 [cited by applicant]
US 20200117996A1 · Chang et al. · 2020 [cited by applicant]
US 20200335091A1 · Chang · 2020 [cited by examiner]
US 20210375289A1 · Zhu · 2021 [cited by examiner]
US 20220351718A1 · Wu · 2022 [cited by examiner]
Chen, Qian et al. “Controllable Time-Delay Transformer for Real-Time Punctuation Prediction and Disfluency Detection.” ICASSP 2020—2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) (… [cited by examiner]
S.-Y. Chang, R. Prabhavalkar, Y. He, T. N. Sainath and G. Simko, “Joint Endpointing and Decoding with End-to-end Models,” ICASSP 2019—2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP… [cited by examiner]
Z. Zhou, T. Tan and Y. Qian, “Punctuation Prediction for Streaming On-Device Speech Recognition,” ICASSP 2022—2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Singapore, Singapore,… [cited by examiner]
Paria Jamshid Lou and Mark Johnson. 2020. Improving Disfluency Detection by Self-Training a Self-Attentive Model. In Proceedings of the 58th Annual Meeting of the Association for Computational Linguistics, pp. 3754-3763… [cited by examiner]
Nguyen, T.S., Pham, N., Stueker, S., & Waibel, A.H. (2020). High Performance Sequence-to-Sequence Model for Streaming Speech Recognition. ArXiv, abs/2003.10022. (Year: 2020). [cited by examiner]
T. Kourkounakis, A. Hajavi and A. Etemad, “FluentNet: End-to-End Detection of Stuttered Speech Disfluencies With Deep Learning,” in IEEE/ACM Transactions on Audio, Speech, and Language Processing, vol. 29, pp. 2986-2999… [cited by examiner]
L. Lu, J. Li and Y. Gong, “Endpoint Detection for Streaming End-to-End Multi-Talker ASR,” ICASSP 2022—2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), Singapore, Singapore, 2022, p… [cited by examiner]
Paria Jamshid Lou et al, “End-to-End Speech Recognition and Disfluency Removal”, arxiv.org,Sep. 23, 2020 (Sep. 23, 2020), p. 1-10, XP081768441. [cited by applicant]
Baiyang Liu et al, “Accurate Endpointing with Expected Pause Duration”, Dresden, Germany Sep. 6, 2015 (Sep. 6, 2015), p. 2912-2916, Interspeech 2015, Retrieved from the Internet: URL:http://www.isca-speech.org/archive/i… [cited by applicant]
International Search Report and Written Opinion issued in related PCT Application No. PCT/US2022/075550, dated Dec. 13, 2022. [cited by applicant]