Zte Corporation

Other

Xi'an Zhongxing New Software Co., Ltd.

Shenzhen Royole Technologies Co., Ltd.

Huawei Device Co., Ltd.

Hao Yuan

Ke Peng

Jiali Li

Guoming Chen

Changbao Zhu

Dongping Jiang

Xu Guan

Zhibin Lin

Zheng Deng

Kaiwen Liu

Jing Lu

Mofei Liu

Xiaojun Qiu

Dongmei Huang

Yijing Li

Na Han

Ming Wu

Hongyan Tu

Mobile terminal

A method for acquiring the number of modified frames for active sound, and a method and apparatus for voice activity detection are disclosed. Firstly, a first voice activity detection decision result and a second voice activity detection decision result are obtained (), the number of hangover frames for active sound is obtained (), and the number of background noise updates is obtained (), and then the number of modified frames for active sound is calculated according to the first voice activity detection decision result, the number of background noise updates and the number of hangover frames for active sound (), and finally, a voice activity detection decision result of a current frame is calculated according to the number of modified frames for active sound and the second voice activity detection decision result ().

Voice activity modification frame acquiring method, and voice activity detection method and apparatus

The present document discloses a method and apparatus for compensating for a lost frame in a transform domain, comprising: calculating frequency-domain coefficients of a current lost frame using frequency-domain coefficients of one or more frames prior to the current lost frame, and performing frequency-time transform to obtain an initially compensated signal; and performing waveform adjustment, to obtain a compensated signal. Alternatively, extrapolation is performed for all or part of frequency points of the current lost frame using phases and amplitudes of corresponding frequency points of a plurality of previous frames to obtain phases and amplitudes of the corresponding frequency points of the current lost frame, to obtain frequency-domain coefficients of the corresponding frequency points, and frequency-time transform is performed to obtain a compensated signal. The above methods can be selected through a judgment algorithm to compensate for the current lost frame, thereby achieving a better compensation effect.

Method and apparatus for frame loss concealment in transform domain

Provided are a Voice Activity Detection (VAD) method and apparatus. The method includes that: at least one first class feature in a first feature category, at least one second class feature in a second feature category and at least two existing VAD judgment results are acquired, the first class feature and the second class feature are features used for VAD detection (S); and VAD is carried out according to the first class feature, the second class feature and the at least two existing VAD judgment results, to obtain a combined VAD judgment result (S).

Voice activity detection method and apparatus

A signal processing method and device are provided. At least two channel sound signals are acquired, and a frequency-domain audio signal corresponding to each channel sound signal is acquired; beam forming output signals of a beam group corresponding to an audio signal of each frequency point are acquired; an output direction of the beam group is acquired; and time-domain sound signals output after beam forming in the output direction are acquired.

Signal processing method and device

Disclosed are a voice activity detection method and device, wherein the method includes: obtaining a combined VAD judgment result according to the number of continuous active frames, an average total SNR of all sub-bands, a tonal signal flag, and at least two existing VAD judgment results (). The method and device are suitable for voice service, solve the problem of inaccurate VAD detection and realize highly accurate VAD judgment.

Voice activity detection method and device

The present document relates to a voice activity detection (VAD) method and methods used for voice activity detection and apparatus thereof, the VAD method includes: obtaining sub-band signals and spectrum amplitudes of a current frame; computing values of a energy feature and a spectral centroid feature of the current frame according to the sub-band signals; computing a signal to noise ratio parameter of the current frame according to a background noise energy estimated from a previous frame, an energy of SNR sub-bands and a energy feature of the current frame; computing a VAD decision result according to a tonality signal flag, a signal to noise ratio parameter, a spectral centroid feature, and a frame energy feature. The methods and apparatus of the present document can improve the accuracy of non-stationary noise (such as office noise) and music detection.

Voice activity detection method and method used for voice activity detection and apparatus thereof

A parameter estimation method for inactive voice signals and a system thereof and comfort noise generation method and system are disclosed. The method includes: for an inactive voice signal frame, performing time-frequency transform on a sequence of time domain signals containing the inactive voice signal frame to obtain a frequency spectrum sequence, calculating frequency spectrum coefficients according to the frequency spectrum sequence, performing smooth processing on the frequency spectrum coefficients, obtaining a smoothly processed frequency spectrum sequence according to the smoothly processed frequency spectrum coefficients, performing inverse time-frequency transform on the smoothly processed frequency spectrum sequence to obtain a reconstructed time domain signal, and estimating an inactive voice signal parameter according to the reconstructed time domain signal to obtain a frequency spectrum parameter and an energy parameter. With the present solution, it can provide stable background noise parameters in a comfort noise generation system at decoding.

Inactive sound signal parameter estimation method and comfort noise generation method and system

A frame loss compensation method and apparatus for audio signals are disclosed. The method includes: when a first frame immediately following a correctly received frame is lost, judging a frame type of the first lost frame, and when the first lost frame is a non-multi-harmonic frame, calculating MDCT coefficients of the first lost frame by using MDCT coefficients of one or more frames prior to the first lost frame; obtaining an initially compensated signal of the first lost frame according to the MDCT coefficients of the first lost frame; and performing a first class of waveform adjustment on the initially compensated signal of the first lost frame and taking an adjusted time-domain signal as a time-domain signal of the first lost frame. The apparatus includes a frame type judgment module, an MDCT coefficient acquisition module, an initial compensation signal acquisition module and an adjustment module.

Frame loss compensation method and apparatus for voice frame signal

The audio coding method and system of lattice vector quantization is provided in the invention. The method comprises: dividing frequency domain coefficients of an audio signal for which a modified discrete cosine transform (MDCT) has been performed into a plurality of coding sub-bands, and quantizing and coding an amplitude envelope value of each coding sub-band to obtain coded bits of amplitude envelopes; performing bit allocation on each coding sub-band, and performing normalization, quantization and coding respectively on vectors in a low bit coding sub-band with pyramid lattice vector quantization and on vectors in a high bit coding sub-band with sphere lattice vector quantization to obtain coded bits of the frequency domain coefficients; multiplexing and packing the coded bits of the amplitude envelope and the coded bits of the frequency domain coefficients of each coding sub-band, then sending them to a decoding side.

Audio-encoding/decoding method and system of lattice-type vector quantizing

Hierarchical audio coding and decoding method and system and hierarchical audio coding and decoding method for transient signals are provided. In the present invention, by introducing a processing method for transient signal frames in the hierarchical audio coding and decoding methods, a segmented time-frequency transform is performed on the transient signal frames, and then the frequency-domain coefficients obtained by transformation are rearranged respectively within the core layer and within the extended layer, so as to perform the same subsequent coding processes, such as bit allocation, frequency-domain coefficient coding, etc., as those on the steady-state signal frames, thus enhancing the coding efficiency of the transient signal frames and improving the quality of the hierarchical audio coding and decoding.

Hierarchical audio frequency encoding and decoding method and system, hierarchical frequency encoding and decoding method for transient signal

The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a Pframe, obtaining a set of frequencies to be predicted, and for each frequency in the set, using phases and amplitudes of a plurality of frames before a (P&#x2212;1)frame in a MDCT-MDST domain to predict a phase and an amplitude of the Pframe, and using the predicted phase and amplitude to obtain a MDCT coefficient of the Pframe at each corresponding frequency; for a frequency outside the set, using MDCT coefficients of a plurality of frames before the Pframe to calculate a MDCT coefficient value of the Pframe at the frequency; performing an IMDCT for the MDCT coefficients of the Pframe to obtain a time domain signal of the Pframe.

Compensator and compensation method for audio frame loss in modified discrete cosine transform domain

The present invention relates to a method and system for audio encoding and decoding and a method for estimating a noise level, and the method for estimating a noise level in the present invention comprises: estimating a power spectrum of an audio signal to be encoded according to a frequency domain coefficient of the audio signal to be encoded; and estimating a noise level of a zero bit encoding subband audio signal according to the power spectrum obtained by calculating, and this noise level for controlling an energy proportion of noise filling to spectral band replication during decoding; wherein a zero bit encoding subband refers to an encoding subband of which allocated bit number is zero. The present invention can well reconstruct the uncoded frequency domain coefficients.

Method and system for audio encoding and decoding and method for estimating noise level

A method and system for performing services in a server and a client of a client/server architecture are provided, and the method includes: when performing to a preset user decision point during the service performing process, the server interrupting the service performing and initiating a user decision request to a client, wherein the user decision request includes information requesting the user to decide a service performing action after the user decision point; after a user decision information made by the client responding to the user decision request is received, determining the service performing action corresponding to the received user decision information according to the corresponding relationship between the user decision information and a service performing rule, and performing the service according to the determined service performing action. The service performing action or flow can be intervened or influenced by the client through the present invention.

Growing community of inventors

Shenzhen, China

Hao Yuan