Keyword spotting using convolutional neural network for speech recognition in Hindi
- URL: http://arxiv.org/abs/2605.02928v1
- Date: Sun, 26 Apr 2026 21:11:29 GMT
- Title: Keyword spotting using convolutional neural network for speech recognition in Hindi
- Abstract summary: We focus on developing an efficient on-device KWS system tailored for user-specific queries.<n>We employ feature engineering techniques to convert raw audio recordings into Mel Frequency Cepstral Coefficients (MFCCs) as an input for our network.<n>Our CNN-based approach achieves a commendable accuracy rate of 91.79% through rigorous evaluation.
- Score: 0.6359663723794672
- License: http://creativecommons.org/licenses/by-nc-nd/4.0/
- Abstract: In this study, we investigate the application of keyword spotting (KWS) in the domain of Hindi speech recognition, utilizing a dataset comprising 40,000 audio samples. With a sampling rate of 44 kHz and an average duration of 1.9 seconds per sample, we focus on developing an efficient on-device KWS system tailored for user-specific queries. Leveraging Convolutional Neural Networks (CNNs) for classification, we employ feature engineering techniques to convert raw audio recordings into Mel Frequency Cepstral Coefficients (MFCCs) as an input for our network. Our experiments encompass various CNN architectures, exploring their efficacy in identifying predefined keywords within the continuous speech stream. Our CNN-based approach achieves a commendable accuracy rate of 91.79% through rigorous evaluation, demonstrating promising performance while ensuring computational efficiency and user-specific customization in Hindi speech recognition.
Related papers
- Quantized Approximate Signal Processing (QASP): Towards Homomorphic Encryption for audio [1.3584036432145363]
Homomorphic encryption (FHE) offers a promising solution by enabling computations on encrypted data and preserving user privacy.<n>Here, we introduce a fully secure pipeline that computes, with FHE and quantized neural network operations.<n>Our methods also support the private computation of audio descriptors and convolutional neural network (CNN) classifiers.
arXiv Detail & Related papers (2025-05-15T17:01:52Z) - Robust Persian Digit Recognition in Noisy Environments Using Hybrid CNN-BiGRU Model [1.5566524830295307]
This study addresses isolated spoken Persian digit recognition (zero to nine) under noisy conditions.<n>A hybrid model combining residual convolutional neural networks and bidirectional gated units (BiGRU) is proposed.<n> Experimental results demonstrate the model's effectiveness, achieving 98.53%, 96.10%, and 95.92% accuracy on training, validation, and test sets.
arXiv Detail & Related papers (2024-12-14T15:11:42Z) - SpokeN-100: A Cross-Lingual Benchmarking Dataset for The Classification of Spoken Numbers in Different Languages [0.0]
Benchmarking plays a pivotal role in assessing and enhancing the performance of compact deep learning models.
Our study introduces a novel, entirely artificially generated benchmarking dataset tailored for speech recognition.
SpokeN-100 consists of spoken numbers from 0 to 99 spoken by 32 different speakers in four different languages.
arXiv Detail & Related papers (2024-03-14T12:07:37Z) - Multilingual Audio-Visual Speech Recognition with Hybrid CTC/RNN-T Fast Conformer [59.57249127943914]
We present a multilingual Audio-Visual Speech Recognition model incorporating several enhancements to improve performance and audio noise robustness.
We increase the amount of audio-visual training data for six distinct languages, generating automatic transcriptions of unlabelled multilingual datasets.
Our proposed model achieves new state-of-the-art performance on the LRS3 dataset, reaching WER of 0.8%.
arXiv Detail & Related papers (2024-03-14T01:16:32Z) - Keyword spotting -- Detecting commands in speech using deep learning [2.709166684084394]
We implement feature engineering by converting raw waveforms to Mel Frequency Cepstral Coefficients (MFCCs)
In our experiments, RNN with BiLSTM and Attention achieves the best performance with an accuracy of 93.9 %.
arXiv Detail & Related papers (2023-12-09T19:04:17Z) - Introducing Semantics into Speech Encoders [91.37001512418111]
We propose an unsupervised way of incorporating semantic information from large language models into self-supervised speech encoders without labeled audio transcriptions.
Our approach achieves similar performance as supervised methods trained on over 100 hours of labeled audio transcripts.
arXiv Detail & Related papers (2022-11-15T18:44:28Z) - Wav2Seq: Pre-training Speech-to-Text Encoder-Decoder Models Using Pseudo
Languages [58.43299730989809]
We introduce Wav2Seq, the first self-supervised approach to pre-train both parts of encoder-decoder models for speech data.
We induce a pseudo language as a compact discrete representation, and formulate a self-supervised pseudo speech recognition task.
This process stands on its own, or can be applied as low-cost second-stage pre-training.
arXiv Detail & Related papers (2022-05-02T17:59:02Z) - On the pragmatism of using binary classifiers over data intensive neural
network classifiers for detection of COVID-19 from voice [34.553128768223615]
We show that detecting COVID-19 from voice does not require custom-made non-standard features or complicated neural network classifiers.
We demonstrate this from a human-curated dataset collected and calibrated in clinical settings.
arXiv Detail & Related papers (2022-04-11T00:19:14Z) - Is Attention always needed? A Case Study on Language Identification from
Speech [1.162918464251504]
The present study introduces convolutional recurrent neural network (CRNN) based LID.
CRNN based LID is designed to operate on the Mel-frequency Cepstral Coefficient (MFCC) characteristics of audio samples.
The LID model exhibits high-performance levels ranging from 97% to 100% for languages that are linguistically similar.
arXiv Detail & Related papers (2021-10-05T16:38:57Z) - On Addressing Practical Challenges for RNN-Transduce [72.72132048437751]
We adapt a well-trained RNN-T model to a new domain without collecting the audio data.
We obtain word-level confidence scores by utilizing several types of features calculated during decoding.
The proposed time stamping method can get less than 50ms word timing difference on average.
arXiv Detail & Related papers (2021-04-27T23:31:43Z) - AutoSpeech: Neural Architecture Search for Speaker Recognition [108.69505815793028]
We propose the first neural architecture search approach approach for the speaker recognition tasks, named as AutoSpeech.
Our algorithm first identifies the optimal operation combination in a neural cell and then derives a CNN model by stacking the neural cell for multiple times.
Results demonstrate that the derived CNN architectures significantly outperform current speaker recognition systems based on VGG-M, ResNet-18, and ResNet-34 back-bones, while enjoying lower model complexity.
arXiv Detail & Related papers (2020-05-07T02:53:47Z) - Deep Speaker Embeddings for Far-Field Speaker Recognition on Short
Utterances [53.063441357826484]
Speaker recognition systems based on deep speaker embeddings have achieved significant performance in controlled conditions.
Speaker verification on short utterances in uncontrolled noisy environment conditions is one of the most challenging and highly demanded tasks.
This paper presents approaches aimed to achieve two goals: a) improve the quality of far-field speaker verification systems in the presence of environmental noise, reverberation and b) reduce the system qualitydegradation for short utterances.
arXiv Detail & Related papers (2020-02-14T13:34:33Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.