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10 years of news and resources for members of the IEEE Signal Processing Society
For our April 2018 issue, we cover recent patents granted in the area of speech coding.
In patent no 9,852,740 a high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.
Patent no. 9,800,453 presents a method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate.
In patent no. 9,767,810 a speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
As presented in patent no. 9,747,916, in a CELP-type speech coding apparatus, switching between an orthogonal search of a fixed codebook and a non-orthogonal search is performed in a practical and effective manner. The CELP-type speech coding apparatus includes a parameter quantizer that selects an adaptive codebook vector and a fixed codebook vector so as to minimize an error between a synthesized speech signal and an input speech signal. The parameter quantizer includes a fixed codebook searcher that switches between the orthogonal fixed codebook search and the non-orthogonal fixed codebook search based on a correlation value between a target vector for the fixed codebook search and the adaptive codebook vector obtained as a result of a synthesis filtering process.
In accordance with one aspect of the invention no. 9,747,915, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.
The invention no. 9,454,972 introduces audio/speech encoding apparatus audio/speech decoding apparatus, audio/speech encoding method and audio/speech decoding method to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, the position of the sub-vector whose codebook indication consumes the most bits is firstly located, and then the value of the codebook is estimated based on the total number of bits available and the bits usage information for other sub-vectors. The difference value is calculated between the actual value and estimated value. Finally, instead of transmitting the codebook indication which consumes the most bits, the position of the sub-vector whose codebook indication consumes the most bits and the difference value between the actual value and the estimated value are transmitted. By applying of the invented method, bits can be saved by the codebook indications.
In patent no. 9,336,790 a speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
Patent no. 9,263,051 presents a method, system and program for decoding a speech signal. In some embodiments, the method comprises: receiving an encoded speech signal having quantization values; transforming the quantization values by adding simulated random-noise samples; and from the encoded speech signal, determining a parameter of the transformation that is usable to control the transformation of the quantization values.
If you have an interesting patent to share when we next feature patents related to speech coding, or if you are especially interested in a signal processing research field that you would want to be highlighted in this section, please send email to Csaba Benedek (benedek.csaba AT sztaki DOT mta DOT hu).
Title: Method for speech coding, method for speech decoding and their apparatuses
Inventors: Yamaura; Tadashi (Tokyo, JP)
Issued: December 26, 2017
Assignee: BlackBerry Limited (Waterloo, Ontario, CA)
Title: Method and apparatus for providing speech coding coefficients using re-sampled coefficients
Inventors: Taleb; Anisse (Stockholm, SE), Xu; Jianfeng (Shenzhen, CN), Virette; David (Munich, DE)
Issued: October 24, 2017
Assignee: Huawei Technologies Co., Ltd., Shenzhen, CN
Title: CELP-type speech coding apparatus and method using adaptive and fixed codebooks
Inventors: Ehara; Hiroyuki (Kanagawa, JP), Hori; Takako (Kanagawa, JP)
Issued: August 29, 2017
Assignee: Panasonic Intellectual Property Corporation of America, Torrance, CA, US
Title: Audio and speech coding device, audio and speech decoding device, method for coding audio and speech, and method for decoding audio and speech
Inventors: Liu; Zongxian (Singapore, SG), Nagisetty; Srikanth (Singapore, SG), Oshikiri; Masahiro (Kanagawa, JP)
Issued: September 27, 2016
Assignee: Panasonic Intellectual Property Corporation of America, Torrance, CA, US
Title: Speech coding by quantizing with random-noise signal
Inventors: Vos; Koen Bernard (San Francisco, CA)
Issued: February 16, 2016
Assignee: Skype (Dublin, IE)
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