What are the common algorithms used in speech recognition?

Speech recognition relies on several core algorithms to convert audio signals into text. These algorithms handle tasks like feature extraction, sequence modeling, and language understanding. While modern systems often combine multiple approaches, three categories of algorithms are particularly foundational: Hidden Markov Models (HMMs), Deep Neural Networks (DNNs), and Connectionist Temporal Classification (CTC). Each addresses specific challenges in processing speech data, such as handling variable-length inputs or capturing temporal dependencies.

HMMs are probabilistic models traditionally used to model sequences of speech sounds. They work by representing phonemes (distinct units of sound) as states, with transitions between states governed by probabilities. For example, an HMM might model the transition from the “s” sound to the “ah” sound in the word “sun.” HMMs are paired with Gaussian Mixture Models (GMMs) to map audio features like Mel-Frequency Cepstral Coefficients (MFCCs) to these states. While HMMs are less dominant today, they laid the groundwork for handling time-series data in speech. Early systems like CMU Sphinx used HMM-GMM hybrids for tasks like digit recognition.

Deep Neural Networks (DNNs), especially Convolutional Neural Networks (CNNs) and Recurrent Neural Networks (RNNs), have become central to modern speech recognition. CNNs process spectrogram-like audio features to detect local patterns (e.g., formants or plosives), while RNNs like Long Short-Term Memory (LSTM) networks model temporal relationships in sequences. For instance, Baidu’s DeepSpeech uses CNNs for initial feature extraction and LSTMs to capture context across time steps. Transformers, with self-attention mechanisms, have also gained traction for their ability to model long-range dependencies. These DNN-based approaches automate feature engineering, reducing reliance on handcrafted acoustic models and improving accuracy on diverse accents or noisy environments.

Connectionist Temporal Classification (CTC) addresses the challenge of aligning audio frames with text outputs. CTC allows models to output characters directly without requiring strict frame-level alignment, which is critical for training on unsegmented data. For example, a CTC loss function might enable a model to output “cat” even if the “c,” “a,” and “t” sounds aren’t perfectly aligned to specific time steps. This is often combined with language models like n-grams or BERT to refine predictions using contextual knowledge (e.g., correcting “recognize speech” instead of “wreck a nice beach”). Open-source tools like Mozilla’s DeepSpeech or OpenAI’s Whisper integrate these components into end-to-end pipelines, balancing acoustic and linguistic modeling for robust performance.