Signal processing method and device
    221.
    发明授权

    公开(公告)号:US10186271B2

    公开(公告)日:2019-01-22

    申请号:US15686175

    申请日:2017-08-25

    Abstract: Present disclosure pertains to an audio signal processing method and device. A current frame of a frequency-domain audio signal has N frequency sub-bands. According to an energy attribute and a spectral attribute of a first subset of M sub-bands, whether to adjust envelope values of the M sub-bands is determined. Based on a determination that the energy envelope values of the M sub-bands need to be adjusted, the energy envelope values of the M sub-bands are modified individually to obtain modified envelope values of the M sub-bands. Encoding bits are allocated to each of the N sub-bands according to the adjusted envelope values of the M sub-bands and energy envelope information of K sub-bands of a second subset. The first subset and the second subset have no overlap in frequency, and N=M+K.

    SPEECH/AUDIO SIGNAL PROCESSING METHOD AND CODING APPARATUS

    公开(公告)号:US20180336910A1

    公开(公告)日:2018-11-22

    申请号:US16051139

    申请日:2018-07-31

    CPC classification number: G10L19/265 G10L19/012 G10L19/18 G10L19/22

    Abstract: The present disclosure provides a speech/audio signal processing method based on wideband switching and a coding apparatus. The method includes: if a first wideband speech/audio signal is a harmonic signal, adjusting a determining condition for determining that a second wideband speech/audio signal is a harmonic signal, to obtain a first determining condition, where the first wideband speech/audio signal is a signal before wideband switching, and the second wideband speech/audio signal is a signal after the wideband switching; and determining, according to the first determining condition, whether the second wideband speech/audio signal is a harmonic signal. In the case of wideband switching, signal types of speech/audio signals remain as consistent as possible before and after the switching, so that continuity of the speech/audio signal decoded by a decoder device is ensured as much as possible, further improving speech communication service quality.

    Frequency envelope vector quantization method and apparatus

    公开(公告)号:US10032460B2

    公开(公告)日:2018-07-24

    申请号:US15715179

    申请日:2017-09-26

    CPC classification number: G10L19/038 G10L19/06 G10L2019/0005

    Abstract: Embodiments of the present application proposes a frequency envelope vector quantization method and apparatus, where the method includes: dividing N frequency envelopes in one frame into N1 vectors; quantizing a first vector in the N1 vectors by using a first codebook, to obtain a code word corresponding to the quantized first vector, where the first codebook is divided into 2B1 portions; determining, according to the code word corresponding to the quantized first vector; determining a second codebook according to the codebook of the ith portion; and quantizing a second vector in the N1 vectors based on the second codebook. In the embodiments of the present application, vector quantization can be performed on frequency envelope vectors by using a codebook with a smaller quantity of bits. Therefore, complexity of vector quantization can be reduced, and an effect of vector quantization can also be ensured.

    CODING/DECODING METHOD, APPARATUS, AND SYSTEM FOR AUDIO SIGNAL

    公开(公告)号:US20170372715A1

    公开(公告)日:2017-12-28

    申请号:US15696591

    申请日:2017-09-06

    CPC classification number: G10L19/12 G10L19/0204 G10L19/0208 G10L19/26

    Abstract: Embodiments of the present application provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.

    METHOD AND APPARATUS FOR PROCESSING TEMPORAL ENVELOPE OF AUDIO SIGNAL, AND ENCODER

    公开(公告)号:US20170098451A1

    公开(公告)日:2017-04-06

    申请号:US15372130

    申请日:2016-12-07

    Inventor: Zexin Liu Lei Miao

    Abstract: A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder are disclosed. When multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes: obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal; dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer, M is greater than or equal to 2; calculating a temporal envelope of each of the subframes; performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using an asymmetric window function; and performing windowing on a subframe except the first subframe and the last subframe of the M subframes.

    Method and apparatus for allocating bits of audio signal
    230.
    发明授权
    Method and apparatus for allocating bits of audio signal 有权
    用于分配音频信号位的方法和装置

    公开(公告)号:US09530420B2

    公开(公告)日:2016-12-27

    申请号:US14675031

    申请日:2015-03-31

    CPC classification number: G10L19/002 G10L19/0204 G10L19/032 G10L19/035

    Abstract: A method and an apparatus for allocating bits of an audio signal. The method includes dividing a frequency band of an audio signal into multiple sub-bands, and quantizing a sub-band normalization factor of each sub-band; classifying the multiple sub-bands into multiple groups, and acquiring a sum of intra-group sub-band normalization factors of each group; performing initial inter-group bit allocation to determine the initial number of bits of each group; performing secondary inter-group bit allocation to allocate coding bits of the audio signal to at least one group; and allocating the bits of the audio signal to sub-bands in the group. The present invention can, by means of grouping, ensure relatively stable allocation in a previous frame and a next frame and reduce an impact of global allocation on local discontinuity in a case of low and medium bit rates.

    Abstract translation: 一种用于分配音频信号的位的方法和装置。 该方法包括将音频信号的频带划分成多个子带,并量化每个子带的子带归一化因子; 将多个子带分为多个组,并获取每组的组内子带归一化因子之和; 执行初始组间比特分配以确定每组的初始比特数; 执行次要组间比特分配以将音频信号的编码比特分配给至少一个组; 并将音频信号的比特分配给组中的子带。 本发明可以通过分组确保在先前帧和下一帧中的相对稳定的分配,并且在低和中比特率的情况下减少全局分配对局部不连续性的影响。

Patent Agency Ranking