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    Speech Recognition and Acoustic Models
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    Research and Experiments

    • Continuous Speech Recognition with a TF-IDF Acoustic Model [View Experiment]
    • Robust automatic speech recognition using acoustic model adaptation prior to missing feature reconstruction [View Experiment]
    • Context-independent Acoustic Models for Thai Speech Recognition [View Experiment]
    • Cross-lingual acoustic model adaptation for speaker-independent speech recognition [View Experiment]
    • Crosslingual acoustic model development for automatic speech recognition (thesis) [View Experiment]
    Background Information

    Definitions

    Speech recognition is a technique that converts spoken words into text.

    An acoustic model is a technique that samples audio recordings of speech, and their text transcriptions, and using software to create statistical representations of the sounds that make up each word. It is used by a speech recognition engine to recognize speech.

    Speech Recognition Applications

    Speech recognition applications include voice user interfaces such as voice dialing (e.g., "Call home"), call routing (e.g., "I would like to make a collect call"), domotic appliance control, search (e.g., find a podcast where particular words were spoken), simple data entry (e.g., entering a credit card number), preparation of structured documents (e.g., a radiology report), speech-to-text processing (e.g., word processors or emails), and aircraft (usually termed Direct Voice Input).

    One of the most notable domains for the commercial application of speech recognition in the United States has been health care and in particular the work of the medical transcriptionist (MT). According to industry experts, at its inception, speech recognition (SR) was sold as a way to completely eliminate transcription rather than make the transcription process more efficient, hence it was not accepted. It was also the case that SR at that time was often technically deficient. Additionally, to be used effectively, it required changes to the ways physicians worked and documented clinical encounters, which many if not all were reluctant to do. The biggest limitation to speech recognition automating transcription, however, is seen as the software. The nature of narrative dictation is highly interpretive and often requires judgment that may be provided by a real human but not yet by an automated system. Another limitation has been the extensive amount of time required by the user and/or system provider to train the software.

    Substantial efforts have been devoted in the last decade to the test and evaluation of speech recognition in fighter aircraft. Of particular note is the U.S. program in speech recognition for the Advanced Fighter Technology Integration (AFTI)/F-16 aircraft (F-16 VISTA), and a program in France installing speech recognition systems on Mirage aircraft, and also programs in the UK dealing with a variety of aircraft platforms. In these programs, speech recognizers have been operated successfully in fighter aircraft, with applications including: setting radio frequencies, commanding an autopilot system, setting steer-point coordinates and weapons release parameters, and controlling flight displays.

    Battle Management command centres generally require rapid access to and control of large, rapidly changing information databases. Commanders and system operators need to query these databases as conveniently as possible, in an eyes-busy environment where much of the information is presented in a display format. Human-machine interaction by voice has the potential to be very useful in these environments. A number of efforts have been undertaken to interface commercially available isolated-word recognizers into battle management environments. In one feasibility study speech recognition equipment was tested in conjunction with an integrated information display for naval battle management applications. Users were very optimistic about the potential of the system, although capabilities were limited.

    Training for air traffic controllers (ATC) represents an excellent application for speech recognition systems. Many ATC training systems currently require a person to act as a "pseudo-pilot", engaging in a voice dialog with the trainee controller, which simulates the dialog which the controller would have to conduct with pilots in a real ATC situation. Speech recognition and synthesis techniques offer the potential to eliminate the need for a person to act as pseudo-pilot, thus reducing training and support personnel. In theory, Air controller tasks are also characterized by highly structured speech as the primary output of the controller, hence reducing the difficulty of the speech recognition task should be possible. In practice this is rarely the case. The FAA document 7110.65 details the phrases that should be used by air traffic controllers. While this document gives less than 150 examples of such phrases, the number of phrases supported by one of the simulation vendors speech recognition systems is in excess of 500,000.

    Speech Recognition Algorithms

    Both acoustic modeling (see below) and language modeling (see below) are important parts of modern statistically-based speech recognition algorithms. Hidden Markov models (HMMs) are widely used in many systems. Language modeling has many other applications such as smart keyboard and document classification.

    Hidden Markov Model: Modern general-purpose speech recognition systems are based on Hidden Markov Models. These are statistical models which output a sequence of symbols or quantities. HMMs are used in speech recognition because a speech signal can be viewed as a piecewise stationary signal or a short-time stationary signal. In a short-time (e.g., 10 milliseconds)), speech can be approximated as a stationary process. Speech can be thought of as a Markov model for many stochastic purposes.

    A hidden Markov model (HMM) is a statistical Markov model in which the system being modeled is assumed to be a Markov process with unobserved (hidden) states. An HMM can be considered as the simplest dynamic Bayesian network.

    Dynamic time warping is an approach that was historically used for speech recognition but has now largely been displaced by the more successful HMM-based approach. Dynamic time warping is an algorithm for measuring similarity between two sequences which may vary in time or speed. For instance, similarities in walking patterns would be detected, even if in one video the person was walking slowly and if in another they were walking more quickly, or even if there were accelerations and decelerations during the course of one observation. DTW has been applied to video, audio, and graphics – indeed, any data which can be turned into a linear representation can be analyzed with DTW.

    Dynamic time warping (DTW) is an algorithm for measuring similarity between two sequences which may vary in time or speed. For instance, similarities in walking patterns would be detected, even if in one video the person was walking slowly and if in another he or she were walking more quickly, or even if there were accelerations and decelerations during the course of one observation. DTW has been applied to video, audio, and graphics — indeed, any data which can be turned into a linear representation can be analyzed with DTW. A well known application has been automatic speech recognition, to cope with different speaking speeds.

    Acoustic Models

    Speech recognition engines require two types of files to recognize speech. They require an acoustic model, which is created by taking audio recordings of speech and their transcriptions (taken from a speech corpus), and 'compiling' them into a statistical representations of the sounds that make up each word (through a process called 'training'). They also require a language model or grammar file. A language model is a file containing the probabilities of sequences of words. A grammar is a much smaller file containing sets of predefined combinations of words. Language models are used for dictation applications, whereas grammars are used in desktop command and control or telephony interactive voice response (IVR) type applications.

    Audio can be encoded at different sampling rates (i.e. samples per second - the most common being: 8 kHz, 16 kHz, 32 kHz, 44.1 kHz, 48 kHz and 96 kHz), and different bits per sample (the most common being: 8-bits, 16-bits or 32-bits). Speech recognition engines work best if the acoustic model they use was trained with speech audio which was recorded at the same sampling rate/bits per sample as the speech being recognized.

    The limiting factor for telephony based speech recognition is the bandwidth at which speech can be transmitted. For example, your standard land-line telephone only has a bandwidth of 64 kbit/s at a sampling rate of 8 kHz and 8-bits per sample (8000 samples per second * 8-bits per sample = 64000 bit/s). Therefore, for telephony based speech recognition, you need acoustic models trained with 8 kHz/8-bit speech audio files.

    Topics of Interest

    A statistical language model assigns a probability to a sequence of m words by means of a probability distribution.

    Language modeling is used in many natural language processing applications such as speech recognition, machine translation, part-of-speech tagging, parsing and information retrieval.

    In speech recognition and in data compression, such a model tries to capture the properties of a language, and to predict the next word in a speech sequence.

    When used in information retrieval, a language model is associated with a document in a collection. With query Q as input, retrieved documents are ranked based on the probability that the document's language model would generate the terms of the query, P(Q|Md). The method to use language models in information retrieval is the query likelihood model.

    In practice, unigram language models are most commonly used in information retrieval, as they are sufficient to determine the topic from a piece of text. Unigram models only calculate the probability of hitting an isolated word, without considering any influence from the words before or after the target. This leads to the Bag of words model, and turns out to generate a multinomial distribution over words.

    For more information:
    http://en.wikipedia.org/wiki/Speech_recognition
    http://en.wikipedia.org/wiki/Acoustic_Model

    Source: Wikipedia (All text is available under the terms of the GNU Free Documentation License and Creative Commons Attribution-ShareAlike License.)

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