METHOD AND SYSTEM FOR DECODING LEFT AND RIGHT CHANNELS OF A STEREO SOUND SIGNAL
    1.
    发明申请
    METHOD AND SYSTEM FOR DECODING LEFT AND RIGHT CHANNELS OF A STEREO SOUND SIGNAL 审中-公开
    用于解码立体声信号的左右声道的方法和系统

    公开(公告)号:WO2017049399A1

    公开(公告)日:2017-03-30

    申请号:PCT/CA2016/051108

    申请日:2016-09-22

    Abstract: A stereo sound decoding method and system decode left and right channels of a stereo sound signal, using received encoding parameters comprising encoding parameters of a primary channel, encoding parameters of a secondary channel, and a factor β. The primary channel encoding parameters comprise LP filter coefficients of the primary channel. The primary channel is decoded in response to the primary channel encoding parameters. The secondary channel is decoded using one of a plurality of coding models, wherein at least one of the coding models uses the primary channel LP filter coefficients to decode the secondary channel. The decoded primary and secondary channels are time domain up-mixed using the factor β to produce the decoded left and right channels of the stereo sound signal, wherein the factor β determines respective contributions of the primary and secondary channels upon production of the left and right channels.

    Abstract translation: 立体声解码方法和系统对立体声信号的左声道和右声道进行解码,使用接收到的编码参数,包括主信道的编码参数,二级信道的编码参数和因子β。 主信道编码参数包括主信道的LP滤波器系数。 响应于主信道编码参数对主信道进行解码。 使用多个编码模型中的一个解码辅助信道,其中至少一个编码模型使用主信道LP滤波器系数来解码辅助信道。 解码的一次和二次信道是使用因子β进行时域上混合,以产生立体声信号的解码的左声道和右声道,其中因子β确定左右生成时主要和次要信道的各自贡献 通道。

    METHOD AND SYSTEM FOR ENCODING A STEREO SOUND SIGNAL USING CODING PARAMETERS OF A PRIMARY CHANNEL TO ENCODE A SECONDARY CHANNEL
    2.
    发明申请
    METHOD AND SYSTEM FOR ENCODING A STEREO SOUND SIGNAL USING CODING PARAMETERS OF A PRIMARY CHANNEL TO ENCODE A SECONDARY CHANNEL 审中-公开
    使用主通道的编码参数编码立体声信号的方法和系统来编辑二次通道

    公开(公告)号:WO2017049398A1

    公开(公告)日:2017-03-30

    申请号:PCT/CA2016/051107

    申请日:2016-09-22

    Abstract: A stereo sound encoding method and system for encoding left and right channels of a stereo sound signal, down mix the left and right channels of the stereo sound signal to produce primary and secondary channels, encode the primary channel, and encode the secondary channel. Encoding the secondary channel comprises analyzing coherence between coding parameters calculated during the secondary channel encoding and coding parameters calculated during the primary channel encoding to decide if the coding parameters calculated during the primary channel encoding are sufficiently close to the coding parameters calculated during the secondary channel encoding to be re-used during the secondary channel encoding.

    Abstract translation: 用于对立体声信号的左右声道进行编码的立体声声音编码方法和系统,将立体声声音信号的左右声道混合,产生主声道和次声道,对主声道进行编码,对第二声道进行编码。 对次要信道的编码包括分析在次信道编码期间计算的编码参数之间的相干性以及在主信道编码期间计算的编码参数之间的相干性,以确定在主信道编码期间计算的编码参数是否足够接近次信道编码期间计算的编码参数 在次要信道编码期间被重新使用。

    METHOD AND DEVICE FOR CLASSIFICATION OF UNCORRELATED STEREO CONTENT, CROSS-TALK DETECTION, AND STEREO MODE SELECTION IN A SOUND CODEC

    公开(公告)号:WO2022051846A1

    公开(公告)日:2022-03-17

    申请号:PCT/CA2021/051238

    申请日:2021-09-08

    Abstract: The present disclosure describes the classification of uncorrelated stereo content (hereinafter "UNCLR classification") and the cross-talk detection (hereinafter "XT ALK detection") in an input stereo sound signal. The present disclosure also describes the stereo mode selection, for example an automatic LRTD/DFT stereo mode selection. Additionally, the disclosure uses said classification so as to select one of a first stereo mode and a second stereo mode for coding a stereo sound signal including a left channel and a right channel; detect cross-talk in a stereo sound signal including a left channel and a right channel in response to features extracted from the stereo sound signal including the left and right channels; or classify of uncorrelated stereo content in a stereo sound signal including a left channel and a right channel in response to features extracted from the stereo sound signal including the left and right channels.

    METHOD AND SYSTEM FOR TIME DOMAIN DOWN MIXING A STEREO SOUND SIGNAL INTO PRIMARY AND SECONDARY CHANNELS USING DETECTING AN OUT-OF-PHASE CONDITION OF THE LEFT AND RIGHT CHANNELS
    4.
    发明申请
    METHOD AND SYSTEM FOR TIME DOMAIN DOWN MIXING A STEREO SOUND SIGNAL INTO PRIMARY AND SECONDARY CHANNELS USING DETECTING AN OUT-OF-PHASE CONDITION OF THE LEFT AND RIGHT CHANNELS 审中-公开
    使用检测左右通道的相位条件的时域降频混合立体声信号到主通道和二次通道的方法和系统

    公开(公告)号:WO2017049396A1

    公开(公告)日:2017-03-30

    申请号:PCT/CA2016/051105

    申请日:2016-09-22

    Abstract: A method and system are implemented in a stereo sound signal encoding system for time domain down mixing right and left channels of an input stereo sound signal into primary and secondary channels. Correlation of the primary and secondary channels of previous frames is determined, and an out-of-phase condition of the left and right channels is detected based on the correlation of the primary and secondary channels of the previous frames. The left and right channels are time domain down mixed, as a function of the detection, to produce the primary and secondary channels using a factor β , wherein the factor β determines respective contributions of the left and right channels upon production of the primary and secondary channels.

    Abstract translation: 在立体声信号编码系统中实现一种方法和系统,用于将输入立体声声音信号的左右声道的时域向下混合成主要和次要信道。 确定先前帧的主信道和次信道的相关性,并且基于先前帧的主信道和次信道的相关性来检测左和右信道的异相状态。 作为检测的函数,左,右通道是时域下混合,以使用因子β来产生主通道和次通道,其中因子β确定在生产主和次级通道时左和右通道的各自贡献 通道。

    DEVICE AND METHOD FOR REDUCING QUANTIZATION NOISE IN A TIME-DOMAIN DECODER
    5.
    发明申请
    DEVICE AND METHOD FOR REDUCING QUANTIZATION NOISE IN A TIME-DOMAIN DECODER 审中-公开
    用于减少时域解码器中的量化噪声的装置和方法

    公开(公告)号:WO2014134702A1

    公开(公告)日:2014-09-12

    申请号:PCT/CA2014/000014

    申请日:2014-01-09

    Abstract: The present disclosure relates to a device and method for reducing quantization noise in a signal contained in a time-domain excitation decoded by a time-domain decoder. The decoded time-domain excitation is converted into a frequency-domain excitation. A weighting mask is produced for retrieving spectral information lost in the quantization noise. The frequency- domain excitation is modified to increase spectral dynamics by application of the weighting mask. The modified frequency-domain excitation is converted into a modified time-domain excitation. The method and device can be used for improving music content rendering of linear-prediction (LP) based codecs. Optionally, a synthesis of the decoded time-domain excitation may be classified into one of a first set of excitation categories and a second set of excitation categories, the second set including INACTIVE or UNVOICED categories, the first set including an OTHER category.

    Abstract translation: 本公开涉及用于减少由时域解码器解码的时域激励中包含的信号中的量化噪声的装置和方法。 解码的时域激发被转换成频域激励。 产生用于检索在量化噪声中丢失的光谱信息的加权掩码。 通过应用加权掩模来修改频域激励以增加光谱动力学。 修改的频域激发被转换成修改的时域激励。 该方法和装置可用于改进基于线性预测(LP)的编解码器的音乐内容呈现。 可选地,解码的时域激励的合成可以被分类为第一组激励类别和第二组激励类别中的一个,第二组包括INACTIVE或UNVOICED类别,第一组包括OTHER类别。

    IMPROVING NON-SPEECH CONTENT FOR LOW RATE CELP DECODER
    6.
    发明申请
    IMPROVING NON-SPEECH CONTENT FOR LOW RATE CELP DECODER 审中-公开
    改进低速CELP解码器的非语音内容

    公开(公告)号:WO2013063688A1

    公开(公告)日:2013-05-10

    申请号:PCT/CA2012/001011

    申请日:2012-11-01

    Abstract: A method and device for modifying a synthesis of a time-domain excitation decoded by a time-domain decoder, wherein the synthesis of the decoded time- domain excitation is classified into one of a number of categories. The decoded time-domain excitation is converted into a frequency-domain excitation, and the frequency-domain excitation is modified as a function of the category in which the synthesis of the decoded time-domain excitation is classified. The modified frequency-domain excitation is converted into a modified time-domain excitation, and a synthesis filter is supplied with the modified time-domain excitation to produce a modified synthesis of the decoded time-domain excitation.

    Abstract translation: 一种用于修改由时域解码器解码的时域激励的合成的方法和装置,其中解码的时域激励的合成被分类为多个类别之一。 解码的时域激励被转换为频域激励,并且频域激励被修改为将解码的时域激励的合成分类的类别的函数。 将修正的频域激励转换为修正的时域激励,并对合成滤波器进行修改的时域激励,以产生解码时域激励的修正合成。

    CODING GENERIC AUDIO SIGNALS AT LOW BITRATES AND LOW DELAY

    公开(公告)号:WO2012055016A8

    公开(公告)日:2012-05-03

    申请号:PCT/CA2011/001182

    申请日:2011-10-24

    Abstract: A mixed time-domain / frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain / frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated. Corresponding encoder and decoder using the mixed time-domain / frequency-domain coding device are also described.

    SYSTEM AND METHOD FOR ENHANCING A DECODED TONAL SOUND SIGNAL

    公开(公告)号:WO2009109050A8

    公开(公告)日:2009-09-11

    申请号:PCT/CA2009/000276

    申请日:2009-03-05

    Abstract: A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.

    METHOD AND DEVICE FOR EFFICIENT QUANTIZATION OF TRANSFORM INFORMATION IN AN EMBEDDED SPEECH AND AUDIO CODEC
    9.
    发明申请
    METHOD AND DEVICE FOR EFFICIENT QUANTIZATION OF TRANSFORM INFORMATION IN AN EMBEDDED SPEECH AND AUDIO CODEC 审中-公开
    嵌入式语音和音频编解码器中的变换信息的有效定量的方法和设备

    公开(公告)号:WO2009039645A1

    公开(公告)日:2009-04-02

    申请号:PCT/CA2008/001700

    申请日:2008-09-25

    CPC classification number: G10L19/12 G10L19/032 G10L19/24

    Abstract: A method and device for coding an input sound signal in at least one lower layer and at least one upper layer of an embedded codec while reducing a quantization noise comprises, in the at least one lower layer, coding the input sound signal to produce coding parameters, wherein coding the input sound signal comprises producing a synthesized sound signal. An error signal is computed as a difference between the input sound signal and the synthesized sound signal and a spectral mask is calculated as a function of a spectrum related to the input sound signal. In the at least one upper layer, the error signal is coded to produce coding coefficients, the spectral mask is applied to the coding coefficients, and the masked coding coefficients are quantized. Applying the spectral mask to the coding coefficients reduces the quantization noise produced upon quantizing the coding coefficients. Therefore, a method and device for reducing the quantization noise produced during coding of the error signal in the at least one upper layer comprises providing the spectral mask and, in the at least one upper layer, applying the spectral mask to the coding coefficients prior to quantizing the coding coefficients.

    Abstract translation: 一种用于在减少量化噪声的同时,在至少一个下层和至少一个嵌入式编解码器的上层编码输入声音信号的方法和装置包括:在至少一个下层中编码输入声音信号以产生编码参数 其中编码所述输入声音信号包括产生合成声音信号。 误差信号被计算为输入声音信号和合成声音信号之间的差,并且根据与输入声音信号相关的频谱的函数计算频谱屏蔽。 在至少一个上层中,对误差信号进行编码以产生编码系数,将频谱掩模应用于编码系数,并对掩蔽的编码系数进行量化。 将频谱掩模应用于编码系数减少了量化编码系数时产生的量化噪声。 因此,用于降低在至少一个上层中的误差信号的编码期间产生的量化噪声的方法和装置包括提供频谱掩模,并且在至少一个上层中,将频谱掩模应用于 量化编码系数。

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