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Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.


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Patent
Bishnu S. Atal1
19 Aug 1968
TL;DR: In this article, an adaptive predictor is employed which is readjusted periodically to match the time-varying characteristics of a speech signal, which is used to reduce the channel capacity required to transmit a signal with specified fidelity.
Abstract: Predictive coding of signals, i.e., the reduction or redundancy in a signal by subtracting from it that part which can be predicted from its past, is a well-known technique for reducing the channel capacity required to transmit a signal with specified fidelity. It has been widely applied to signals, such as television signals which have regularly repeating intervals of information, but has not been satisfactorily applied to signals, such as speech, which exhibit characteristics that vary from speaker to speaker and from time to time for one speaker. According to this invention, an adaptive predictor is employed which is readjusted periodically to match the time-varying characteristics of a speech signal.

181 citations

Journal ArticleDOI
TL;DR: It is found that lossless audio coders have reached a limit in what can be achieved for lossless compression of audio, and a new lossless Audio coder is described called AudioPak, which low algorithmic complexity and performs well or even better than most of the losslessaudio coders that have been described in the literature.
Abstract: Lossless audio compression is likely to play an important part in music distribution over the Internet, DVD audio, digital audio archiving, and mixing. The article is a survey and a classification of the current state-of-the-art lossless audio compression algorithms. This study finds that lossless audio coders have reached a limit in what can be achieved for lossless compression of audio. It also describes a new lossless audio coder called AudioPak, which low algorithmic complexity and performs well or even better than most of the lossless audio coders that have been described in the literature.

181 citations

PatentDOI
TL;DR: In this paper, a speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter applied computation to the drive signal vector in the pitch forecast called either closed loop or compatible code book is minimized.
Abstract: This invention provides a novel speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toeplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter-applied computation to the drive signal vector in the pitch forecast called either "closed loop" or "compatible code book" is minimized. The vector quantization method substantially making up the speech coding system of the invention is characteristically used by the system.

181 citations

Patent
11 Dec 2001
TL;DR: In this paper, a computer implemented method to map an audio file to a verbatim text file is described, which includes a computer-implemented method for mapping audio files to text files.
Abstract: The invention includes a computer implemented method to map an audio file to a verbatim text file. First, a first window may be loaded with a transcribed text file having words. The transcribed text file is associated with the audio file. A second window is loaded with the verbatim text file having a plurality of words. At least one word from the transcribed text file and at least one word from the verbatim text file are selected. The at least one word from the transcribed text file and the at least one word from the verbatim text file is linked. The steps are repeated until all the words in the verbatim text file have been linked.

179 citations

Patent
27 Mar 1995
TL;DR: A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component as mentioned in this paper.
Abstract: Apparatus and methods for including a code having at least one code frequency component in an audio signal are provided. The abilities of various frequency components in the audio signal to mask the code frequency component to human hearing are evaluated and based on these evaluations an amplitude is assigned to the code frequency component. Methods and apparatus for detecting a code in an encoded audio signal are also provided. A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component.

179 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202338
202284
202170
202062
201977
2018108