Glossary
Hidden Markov model (HMM)
A hidden Markov model (HMM) is a statistical model in which a sequence of hidden states produces the observations you can measure, and each state changes to the next with set probabilities. In speech recognition, the hidden states are units of sound and the observations are short frames of audio.
Updated October 17, 20263 cited facts, each traced to its source
Also called HMM.
A hidden Markov model (HMM) is a statistical model with two layers of data. The hidden states follow one another with set transition probabilities, and each state produces an observation you can measure. You see the observations, not the states, so the states are "hidden". Rabiner's 1989 tutorial lays out the three problems an HMM must solve before it is useful for speech: evaluating how likely an observation sequence is, finding the best sequence of states, and adjusting the model's parameters to fit the signal 1.
In speech, the hidden states are units of sound
A word is modeled as a sequence of states, each standing for a small unit of sound. Each short frame of audio is an observation. A scoring model estimates how well a given state fits a frame. Older systems used a Gaussian mixture model for that score, and Hinton and colleagues describe the pipeline as HMMs handling the timing of speech while Gaussian mixture models score how well each state fits a short window of acoustic input 2.
The transition probabilities handle timing. Take the "k" sound at the start of "cat". The model stays in that state for as many frames as the sound lasts, then moves on to the next. A slow speaker keeps the model in each state longer, and a fast speaker moves through the states more quickly, which is why one word can take a different number of frames each time it is spoken.
Decoding finds the most likely path through the states
To turn audio into words, a decoder searches for the most probable path through the states. The search is usually done with the Viterbi algorithm, which keeps only the best path into each state at each frame. A language model is combined with the search, so that a word which sounds right but makes a poor sentence is less likely to win.
Neural networks improved the scores, but the timing model stayed
In a 2012 review, a deep neural network reduced the word error rate of a GMM-HMM baseline from 27.4 to 18.5 percent on a Switchboard telephone task 3. The result was a turning point. Many systems kept the HMM for timing and replaced the Gaussian mixtures with neural networks.
Later systems, including CTC and RNN transducer models, learn the mapping from audio to text in one network and no longer need a separate HMM. That shift is why older speech documentation talks about acoustic models and decoders, and the newer material often does not.
Why it matters for dictation
Early dictation software was built on HMMs, so older manuals and papers use their vocabulary. Knowing what a hidden state and a transition probability are helps you read those documents and understand why error rates fell so sharply after neural networks arrived. It also explains why a timing model is part of a traditional system, while a modern end-to-end model handles timing inside the network.
If a product describes its acoustic model and language model as separate parts, it is probably following the older design. If it describes one neural network, it is probably not.
Related terms
- Acoustic model: the part that scores sound against speech units.
- Language model (in speech recognition): the part that scores word sequences.
- Connectionist temporal classification (CTC): a training method that replaced the HMM's timing role in many systems.
- Automatic speech recognition (ASR): the field the HMM belongs to.
Guides that cover it
Evidence
Every number on this page, with where it comes from. Open a fact to see the exact words in its source.
- 1.Rabiner's 1989 tutorial says HMMs must solve three fundamental problems before they apply to speech: evaluating the likelihood of an observation sequence, finding the best sequence of model states, and adjusting model parameters to fit the observed signal.
Lawrence R. Rabiner, Proceedings of the IEEE, vol. 77, no. 2, pp. 257-286, 1989 · Original
- 2.
- 3.
Related terms
- Speech Recognition: What It Is and How Well It WorksThe topic this term belongs to
- Acoustic modelAn acoustic model is the part of a speech recognizer that scores how well a short window of audio matches the sounds of speech. Since about 2012 most acoustic models are deep neural networks, and their training data strongly affects accuracy.
- Automatic speech recognition (ASR)Automatic speech recognition (ASR) is software that converts spoken audio into written words. Modern systems are neural networks trained on large amounts of transcribed speech, and they run in the cloud or on the device.
- ConformerA Conformer is a neural encoder for speech recognition that combines self-attention, which sees the whole input, with convolution, which captures local sound patterns. FastConformer is NVIDIA's faster variant of the same design, used in streaming models.
- Connectionist temporal classification (CTC)Connectionist temporal classification (CTC) is a way to train a neural network to turn audio into text without knowing in advance which moment of sound matches which letter. The network labels every short slice of audio with a character or a blank, and the blanks and repeated labels are then removed to leave the transcript.
- Inverse text normalization (ITN)Inverse text normalization (ITN) is the step that converts a speech recognizer's spoken-form output into written form, so that 'two hundred' becomes 200. It changes how the text reads, not which words the recognizer heard.
- Language model (in speech recognition)In speech recognition, a language model (LM) scores how probable a word sequence is in a language. The recognizer combines that score with the acoustic evidence to choose between words that sound alike, such as homophones.
- LibriSpeechLibriSpeech is a free corpus of about 1,000 hours of read English speech, taken from LibriVox audiobooks, and it is one of the most widely used benchmarks for English speech recognition. Its scores describe clean, read speech, which differs from everyday dictation.
- Mel spectrogramA Mel spectrogram is a grid of sound energy over time, with frequency bands spaced to match human hearing. Many speech recognizers compute its logarithm, the log-Mel spectrogram, and feed it to a neural network instead of the raw waveform.
- Real-time factor (RTF)The real-time factor (RTF) is the time a system takes to process audio divided by the length of that audio. An RTF of 1 or lower means the system keeps up with real-time speech. Some leaderboards report the inverse, RTFx, where higher is faster.
- RNN transducer (RNN-T)An RNN transducer (RNN-T) is a neural speech model that reads audio in order and emits words as it goes, so text can appear while the speaker is still talking. It has an encoder for the sound, a prediction network for the words already written, and a joint network that chooses the next output.
- Speaker diarizationSpeaker diarization is the task of working out who spoke when in a recording, by splitting the audio into turns and grouping turns that come from the same voice. It is separate from transcribing the words, and a transcript gains speaker labels only when diarization is run alongside recognition.
- Voice activity detection (VAD)Voice activity detection (VAD) is a component that decides whether a stretch of audio contains speech. Speech recognizers use it to split audio into turns, skip silence and decide when a speaker has finished.
Pages that use this term
- Speech Recognition: What It Is and How Well It WorksTopic
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- Speech Recognition Models: How Neural ASR WorksGuide
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- History of Speech Recognition: From Audrey to WhisperGuide
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- Acoustic modelGlossary
- Automatic speech recognition (ASR)Glossary
- Connectionist temporal classification (CTC)Glossary
- Machine learningGlossary