Audio coding device, method, and computer-readable recording medium storing program
    91.
    发明授权
    Audio coding device, method, and computer-readable recording medium storing program 有权
    音频编码装置,方法和计算机可读记录介质存储程序

    公开(公告)号:US09111533B2

    公开(公告)日:2015-08-18

    申请号:US13297536

    申请日:2011-11-16

    CPC classification number: G10L19/035 G10L19/0017 G10L19/008 G10L19/0204

    Abstract: An audio coding device includes a time-to-frequency converter that performs time-to-frequency conversion on each frame of a signal in at least one channel included in an audio signal in a predetermined length of time in order to convert the signal in the at least one channel to a frequency signal; a complexity calculator that calculates complexity of the frequency signal for each of the at least one channel. The audio further includes a bit allocation controller that determines a number of bits to be allocated to each of at least one channel so that more bits are allocated to the each of the at least one channel as the complexity of the each of at least one channel increases, and increases the number of bits to be allocated as an estimation error in the number; and a coder that codes the frequency signal.

    Abstract translation: 一种音频编码装置包括一个时间 - 频率转换器,它以预定的时间长度在包括在音频信号中的至少一个信道中的信号的每个帧上进行时间 - 频率转换,以便将信号转换成 至少一个频道到频率信号; 复杂度计算器,其计算所述至少一个信道中的每一个的频率信号的复杂度。 音频还包括位分配控制器,其确定要被分配给至少一个信道中的每一个的比特数,使得更多的比特被分配给至少一个信道中的每一个,因为至少一个信道中的每一个的复杂度 增加并增加要分配的比特数作为该数量中的估计误差; 以及编码频率信号的编码器。

    Encoding device and encoding method, decoding device and decoding method, and program
    94.
    发明授权
    Encoding device and encoding method, decoding device and decoding method, and program 有权
    编码装置和编码方法,解码装置及解码方法及程序

    公开(公告)号:US08892429B2

    公开(公告)日:2014-11-18

    申请号:US13583994

    申请日:2011-03-08

    CPC classification number: G10L19/035 G10L19/0212

    Abstract: The present invention relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program that reduce deterioration of sound quality due to encoding of audio signals.An envelope emphasis part (51) emphasizes an envelope (ENV). A noise shaping part (52) divides an emphasized envelope (D) formed by emphasis of the envelope (ENV) by a value larger than 1, and subtracts noise shaping (G) specified by information (NS) from a result of the division. A quantization part (14) sets a result of the subtraction as a quantization bit count (WL), and quantizes a normalized spectrum (S1) formed by normalization of a spectrum (S0) based on the quantization bit count (WL). A multiplexing part (53) multiplexes the information (NS), a quantized spectrum (QS) formed by quantization of the normalized spectrum (S1), and the envelope (ENV). The present invention can be applied to an encoding device encoding audio signals, for example.

    Abstract translation: 编码装置和编码方法,解码装置和解码方法技术领域本发明涉及一种减少音频信号编码导致的音质劣化的程序。 信封重点部分(51)强调信封(ENV)。 噪声整形部分(52)将由包络(ENV)的强调形成的强调包络(D)除以大于1的值,并从分割结果中减去由信息(NS)指定的噪声整形(G)。 量化部分(14)将减法的结果设置为量化位计数(WL),并且通过基于量化位计数(WL)对通过频谱归一化形成的归一化频谱(S1)进行量化。 复用部分(53)复用信息(NS),通过归一化频谱(S1)的量化形成的量化频谱(QS)和信封(ENV)。 例如,本发明可以应用于编码音频信号的编码装置。

    Audio signal signal-to-mask ratio processor for subband coding
    96.
    发明授权
    Audio signal signal-to-mask ratio processor for subband coding 失效
    用于子带编码的音频信号信号到掩码比率处理器

    公开(公告)号:US5832427A

    公开(公告)日:1998-11-03

    申请号:US656476

    申请日:1996-05-31

    CPC classification number: H04B1/665

    Abstract: In an audio signal processing circuit for carrying out a subband coding used for an audio signal coding, a subband filter (12A) receives 1152 audio samples (SI) of each one frame and divides the samples into 32 frequency bands to sequentially output a vector of first half subband signals (SFA) for a first half of the one frame and a vector of second half subband signals (SFB) for a second half of the one frame. A FFT circuit (111A) carries out an FFT processing for 512 audio samples of each of the first half and the second half of each one frame, to sequentially generate a first half power spectrum (PSA) and a second half power spectrum (PSB). A calculating circuit (113A) calculates a first half SMR vector (SMA) on the basis of the first half subband signals (SFA) and the first half power spectrum (PSA), and then, a second half SMR vector (SMB) on the basis of the second half subband signals (SFB) and the second half power spectrum (PSB). A comparing circuit (115) outputs a larger one of the first half SMR vector (SMA) and the second half SMR vector (SMB), as an SMR vector (SM) for the whole of a corresponding one frame.

    Abstract translation: 在用于执行用于音频信号编码的子带编码的音频信号处理电路中,子带滤波器(12A)接收每一帧的1152个音频样本(SI),并将样本划分为32个频带,以顺序地输出 一帧的前半部分的前半个子带信号(SFA)和一帧后半部分的第二半子带信号(SFB)的向量。 FFT电路(111A)对每一帧的前半部分和后半部分中的每一个的512个音频采样执行FFT处理,以顺序生成第一半功率谱(PSA)和第二半功率谱(PSB) 。 计算电路(113A)基于前半个子带信号(SFA)和第一半功率谱(PSA)来计算前半个SMR向量(SMA),然后,在第 第二半子带信号(SFB)和第二半功率谱(PSB)的基础。 比较电路(115)输出第一半SMR矢量(SMA)和第二半SMR矢量(SMB)中较大的一个,作为用于整个相应一帧的SMR矢量(SM)。

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