Speech recognition

Speech recognition, computer speech recognition, or speech-to-text ) is a sub-field of computational linguistics concerned with methods and technologies that translate spoken language into text or other interpretable forms.
Speech recognition applications include voice user interfaces, where the user speaks to a device, which "listens" and processes the audio. Common voice applications include interpreting commands for calling, call routing, home automation, and aircraft control. These applications are called direct voice input. Productivity applications include searching audio recordings, creating transcripts, and dictation.
Speech recognition can be used to analyse speaker characteristics, such as identifying native language using pronunciation assessment.
Voice recognition refers to identifying the speaker, rather than speech contents. Recognizing the speaker can simplify the task of translating speech in systems trained on a specific person's voice. It can also be used to authenticate the speaker as part of a security process.

History

Applications for speech recognition developed over many decades, with progress accelerated due to advances in deep learning and the use of big data. These advances are reflected in an increase in academic papers, and greater system adoption.
Key areas of growth include vocabulary size, more accurate recognition for unfamiliar speakers, and faster processing speed.

Pre-1970

1952 – Bell Labs researchers, Stephen Balashek, R. Biddulph, and K. H. Davis, built Audrey for single-speaker digit recognition. Their system located the formants in the power spectrum of each utterance.
1960 – Gunnar Fant developed and published the source-filter model of speech production.
1962 – IBM's 16-word "Shoebox" machine's speech recognition debuted at the 1962 World's Fair.
1966 – Linear predictive coding, a speech coding method, was proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone.
1969 – Funding at Bell Labs came to a halt for several years after the company's head engineer, John R. Pierce, wrote an open letter criticizing speech recognition research. This defunding lasted until Pierce retired and James L. Flanagan took over.

Raj Reddy was the first person to work on continuous speech recognition, as a graduate student at Stanford University in the late 1960s. Previous systems required users to pause after each word. Reddy's system issued spoken commands for playing chess.
Around this time, Soviet researchers invented the dynamic time warping algorithm and used it to create a recognizer capable of operating on a 200-word vocabulary. DTW processed speech by dividing it into short frames and treating each frame as a unit. Speaker independence, however, remained unsolved.

1970–1990

1971 – DARPA funded a five-year speech recognition research project, Speech Understanding Research, seeking a minimum vocabulary size of 1,000 words. The project considered speech understanding a key to achieving progress in speech recognition, which was later disproved. BBN, IBM, Carnegie Mellon, and Stanford Research Institute participated.
1972 – The IEEE Acoustics, Speech, and Signal Processing group held a conference in Newton, Massachusetts.
1976 – The first ICASSP was held in Philadelphia, which became a major venue for publishing on speech recognition.

During the late 1960s, Leonard Baum developed the mathematics of Markov chains at the Institute for Defense Analysis. A decade later, at CMU, Raj Reddy's students James Baker and Janet M. Baker began using the hidden Markov model for speech recognition. James Baker had learned about HMMs while at the Institute for Defense Analysis. HMMs enabled researchers to combine sources of knowledge, such as acoustics, language, and syntax, in a unified probabilistic model.
By the mid-1980s, Fred Jelinek's team at IBM created a voice-activated typewriter called Tangora, which could handle a 20,000-word vocabulary. Jelinek's statistical approach placed less emphasis on emulating human brain processes in favor of statistical modelling. This was controversial among linguists since HMMs are too simplistic to account for many features of human languages. However, the HMM proved to be a highly useful way for modelling speech and replaced dynamic time warping as the dominant speech recognition algorithm in the 1980s.

1982 – Dragon Systems, founded by James and Janet M. Baker, was one of IBM's few competitors.
Practical speech recognition

The 1980s also saw the introduction of the n-gram language model.

1987 – The back-off model enabled language models to use multiple-length n-grams, and CSELT used HMM to recognize languages.

At the end of the DARPA program in 1976, the best computer available to researchers was the PDP-10 with 4 MB of RAM. It could take up to 100 minutes to decode 30 seconds of speech.
Practical products included:

1984 – the Apricot Portable was released with up to 4096 words support, of which only 64 could be held in RAM at a time.
1987 – a recognizer from Kurzweil Applied Intelligence
1990 – Dragon Dictate, a consumer product released in 1990. AT&T deployed the Voice Recognition Call Processing service in 1992 to route telephone calls without a human operator. The technology was developed by Lawrence Rabiner and others at Bell Labs.

By the early 1990s, the vocabulary of the typical commercial speech recognition system had exceeded the average human vocabulary. Reddy's former student, Xuedong Huang, developed the Sphinx-II system at CMU. Sphinx-II was the first to do speaker-independent, large vocabulary, continuous speech recognition, and it won DARPA's 1992 evaluation. Handling continuous speech with a large vocabulary was a major milestone. Huang later founded the speech recognition group at Microsoft in 1993. Reddy's student Kai-Fu Lee joined Apple, where, in 1992, he helped develop the Casper speech interface prototype.
Lernout & Hauspie, a Belgium-based speech recognition company, acquired other companies, including Kurzweil Applied Intelligence in 1997 and Dragon Systems in 2000. L&H was used in Windows XP. L&H was an industry leader until an accounting scandal destroyed it in 2001. L&H speech technology was bought by ScanSoft, which became Nuance in 2005. Apple licensed Nuance software for its digital assistant Siri.

2000s

In the 2000s, DARPA sponsored two speech recognition programs: Effective Affordable Reusable Speech-to-Text in 2002, followed by Global Autonomous Language Exploitation in 2005. Four teams participated in EARS: IBM; a team led by BBN with LIMSI and the University of Pittsburgh; Cambridge University; and a team composed of ICSI, SRI, and the University of Washington. EARS funded the collection of the Switchboard telephone speech corpus, which contained 260 hours of recorded conversations from over 500 speakers. The GALE program focused on Arabic and Mandarin broadcast news. Google's first effort at speech recognition came in 2007 after recruiting Nuance researchers. Its first product, GOOG-411, was a telephone-based directory service.
Since at least 2006, the U.S. National Security Agency has employed keyword spotting, allowing analysts to index large volumes of recorded conversations and identify speech containing "interesting" keywords. Other government research programs focused on intelligence applications, such as DARPA's EARS program and IARPA's Babel program.
In the early 2000s, speech recognition was dominated by hidden Markov models combined with feed-forward artificial neural networks. Later, speech recognition was taken over by long short-term memory, a recurrent neural network published by Sepp Hochreiter & Jürgen Schmidhuber in 1997. LSTM RNNs avoid the vanishing gradient problem and can learn "Very Deep Learning" tasks that require memories of events that happened thousands of discrete time steps earlier, which is important for speech.
Around 2007, LSTMs trained with Connectionist Temporal Classification began to outperform. In 2015, Google reported a 49 percent error‑rate reduction in its speech recognition via CTC‑trained LSTM. Transformers, a type of neural network based solely on attention, were adopted in computer vision and language modelling, and then to speech recognition.
Deep feed-forward networks for acoustic modelling were introduced in 2009 by Geoffrey Hinton and his students at the University of Toronto, and by Li Deng and colleagues at Microsoft Research. In contrast to the prioer incremental improvements, deep learning decreased error rates by 30%.
Both shallow and deep forms of ANNs had been explored since the 1980s. However, these methods never defeated non-uniform internal-handcrafting Gaussian mixture model/hidden Markov model technology. Difficulties analyzed in the 1990s, included gradient diminishing and weak temporal correlation structure. All these difficulties combined with insufficient training data and computing power. Most speech recognition pursued generative modelling approaches until deep learning won the day. Hinton et al. and Deng et al.

2010s

By early the 2010s, speech recognition was differentiated from speaker recognition, and speaker independence was considered a major breakthrough. Until then, systems required a "training" period for each voice.
In 2017, Microsoft researchers reached the human parity milestone of transcribing conversational speech on the widely benchmarked Switchboard task. Multiple deep learning models were used to improve accuracy. The error rate was reported to be as low as 4 professional human transcribers working together on the same benchmark.

Models, methods, and algorithms

Both acoustic modeling and language modeling are important parts of statistically-based speech recognition algorithms. Hidden Markov models are widely used in many systems. Language modelling is also used in many other natural language processing applications, such as document classification or statistical machine translation.