ASR
How it works
A typical ASR pipeline starts with a feature frontend: the signal is split into short frames and converted into log-mel filterbanks (formerly MFCCs). An acoustic encoder (RNN, Transformer or Conformer) turns the feature sequence into hidden representations. A training objective ties them to text via one of three approaches: CTC marginalises over all alignments using a blank token, RNN-T adds a prediction network and a joint network that emit output token by token, and attention-based encoder-decoder models decode autoregressively. Decoding uses beam search, often with shallow fusion of an external language model. Quality is evaluated with WER.
Problem solved
Speech is a time-varying, continuous signal with no clear word boundaries, and the same utterance varies with accent, speaking rate, noise and recording channel. ASR addresses the problem of mapping such a signal to discrete text without manual audio-to-letter alignment and in a way that is robust to this variability.
Components
Converts the raw signal into feature frames (log-mel filterbanks / MFCCs).
Models the feature sequence into hidden representations (RNN, Transformer, Conformer).
Maps acoustic representations to text tokens (CTC, RNN-T, attention).
Optional external language model improving linguistic correctness during decoding.
Official
Implementation
WER rises sharply with noise, accents, overlapping speech and recording channels unlike the training data.
Encoder-decoder models (e.g. Whisper) can generate text not grounded in the audio, especially on silence or long recordings.
Models trained on short windows (e.g. 30 s in Whisper) require chunking long audio, which can cause boundary errors.
Evolution
Recognition of spoken digits from a single speaker — one of the first ASR systems.
Statistical acoustic models based on hidden Markov models dominate ASR for decades.
Sequence training without manual audio-to-label alignment — a foundation of end-to-end ASR.
Deep neural acoustic models and RNN-T transducers (Graves) as an alternative to CTC.
Large-scale end-to-end ASR based on CTC (Baidu).
Attention-based encoder-decoder model for ASR.
Convolution-augmented Transformer; SOTA results on LibriSpeech.
Self-supervised learning of speech representations reducing the need for labelled data.
Large-scale weakly-supervised training; robust, multilingual ASR (OpenAI).
Hyperparameters (configurable axes)
The network that encodes the acoustic signal.
How acoustics are tied to text and output is produced.
Output tokenization level.
Inclusion of an external language model during decoding.
Input representation of the signal.
Execution paradigm
Modern ASR models (Transformer/Conformer, CTC/RNN-T/attention) are dense neural networks; the whole network is active for each audio frame.
Parallelism
The acoustic encoder (especially Transformer/Conformer, CTC) processes frames in parallel during training, whereas autoregressive decoding (attention, RNN-T) is sequential token by token.
Hardware requirements
Training and batched inference of large acoustic models (Transformer/Conformer) benefit from matrix multiplication on tensor cores.
Lightweight CTC/RNN-T models and on-device streaming ASR run on CPU.