Audio Encoding Device, Audio Decoding Device, and Method Thereof
    21.
    发明申请
    Audio Encoding Device, Audio Decoding Device, and Method Thereof 有权
    音频编码装置,音频解码装置及其方法

    公开(公告)号:US20070250310A1

    公开(公告)日:2007-10-25

    申请号:US11630380

    申请日:2005-06-16

    IPC分类号: G10L19/04

    CPC分类号: G10L19/24 G10L19/12

    摘要: There is disclosed an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal. In this device, a first encoding section (115) subjects an input signal (S11) to audio encoding processing of the CELP method and outputs the obtained first encoded information (S12) to a parameter decoding section (120). The parameter decoding section (120) acquires a first quantization LSP code (L1), a first adaptive excitation lag code (A1), and the like from the first encoded information (S12), obtains a first parameter group (S13) from these codes, and outputs it to a second encoding section (130). The second encoding section (130) subjects the input signal (S11) to a second encoding processing by using the first parameter group (S13) and obtains second encoded information (S14). A multiplexing section (154) multiplexes the first encoded information (S12) with the second encoded information (S14) and outputs them via a transmission path N to a decoding apparatus (150).

    摘要翻译: 公开了一种音频编码装置,当对音频信号进行分层编码时,能够在扩展层中使用CELP方法的音频编码实现有效的编码。 在该装置中,第一编码部(115)使输入信号(S11)进行CELP方式的音频编码处理,并将获得的第一编码信息(S12)输出到参数解码部(120)。 参数解码部(120)从第一编码信息(S12)获取第一量化LSP码(L 1),第一自适应激励滞后码(A 1)等,得到第一参数组(S13 ),并将其输出到第二编码部(130)。 第二编码部(130)通过使用第一参数组(S13)对输入信号(S11)进行第二编码处理,并获得第二编码信息(S14)。 复用部(154)将第一编码信息(S12)与第二编码信息(S14)进行多路复用,并经由传输路径N将其输出到解码装置(150)。

    EXCITATION VECTOR GENERATOR, SPEECH CODER AND SPEECH DECODER
    22.
    发明申请
    EXCITATION VECTOR GENERATOR, SPEECH CODER AND SPEECH DECODER 有权
    激励矢量发生器,语音编码器和语音解码器

    公开(公告)号:US20060235682A1

    公开(公告)日:2006-10-19

    申请号:US11421932

    申请日:2006-06-02

    IPC分类号: G10L19/12

    摘要: An excitation vector generator includes an input vector providing system that is capable of providing an input vector having at least one pulse, each pulse having a predetermined position and a respective polarity. A fixed waveform storage system is capable of storing at least one fixed waveform. An arranging system is capable of arranging the at least one fixed waveform in accordance with the position and the polarity of the at least one pulse.

    摘要翻译: 激励矢量发生器包括输入矢量提供系统,其能够提供具有至少一个脉冲的输入矢量,每个脉冲具有预定位置和相应的极性。 固定波形存储系统能够存储至少一个固定波形。 排列系统能够根据至少一个脉冲的位置和极性排列至少一个固定波形。

    Speech coding apparatus and speech decoding apparatus
    23.
    发明授权
    Speech coding apparatus and speech decoding apparatus 有权
    语音编码装置和语音解码装置

    公开(公告)号:US07110943B1

    公开(公告)日:2006-09-19

    申请号:US09462493

    申请日:1999-06-08

    IPC分类号: G10L19/12

    摘要: First codebook 61 and second codebook 62 respectively have two subcodebooks, and in respective codebooks, addition sections 66 and 67 obtain respective excitation vectors by adding sub-excitation vectors fetched from respective two subcodebooks. Addition section 68 obtains an excitation sample by adding those excitation vectors. According to the aforementioned constitution, it is possible to store sub-excitation vectors with different characteristics in respective sub-codebooks. Therefore, it is possible to correspond to input signals with various characteristics, and achieve excellent sound qualities at the time of decoding.

    摘要翻译: 第一码本61和第二码本62分别具有两个子码本,并且在相应码本中,相加部分66和67通过添加从相应的两个子码本中提取的子激励矢量来获得相应的激励矢量。 加法部分68通过加上这些激励矢量来获得激发样本。 根据上述结构,可以在各子码本中存储具有不同特性的子激励矢量。 因此,可以对应于具有各种特性的输入信号,并且在解码时获得优良的声音质量。

    Audio encoding device, audio decoding device, audio encoding method, and audio decoding method
    24.
    发明申请
    Audio encoding device, audio decoding device, audio encoding method, and audio decoding method 有权
    音频编码装置,音频解码装置,音频编码方法和音频解码方法

    公开(公告)号:US20060173677A1

    公开(公告)日:2006-08-03

    申请号:US10554619

    申请日:2004-04-30

    IPC分类号: G10L19/12

    CPC分类号: G10L19/24

    摘要: Base layer coding section 101 encodes an input signal to obtain base layer coded information. Base layer decoding section 102 decodes the base layer coded information to obtain a base layer decoded signal and long term prediction information (pitch lag). Adding section 103 inverts the polarity of the base layer decoded signal to add to the input signal, and obtains a residual signal. Enhancement layer coding section 104 encodes a long term prediction coefficient calculated using the long term prediction information and the residual signal to obtain enhancement layer coded information. Base layer decoding section 152 decodes the base layer coded information to obtain the base layer decoded signal and long term prediction information. Using the long term prediction information, enhancement layer decoding section 153 decodes the enhancement layer coded information to obtain an enhancement layer decoded signal. Adding section 154 adds the base layer decoded signal and enhancement layer decoded signal to obtain a speech/sound signal. It is thereby possible to implement scalable coding with small amounts of calculation and coded information.

    摘要翻译: 基层编码单元101对输入信号进行编码,得到基本层编码信息。 基层解码单元102对基本层编码信息进行解码,得到基本层解码信号和长期预测信息(音调滞后)。 加法部103将基本层解码信号的极性反转为输入信号,得到残留信号。 增强层编码部分104对使用长期预测信息和残差信号计算的长期预测系数进行编码,以获得增强层编码信息。 基层解码单元152对基本层编码信息进行解码,得到基本层解码信号和长期预测信息。 使用长期预测信息,增强层解码单元153对增强层编码信息进行解码,得到增强层解码信号。 添加部分154添加基本层解码信号和增强层解码信号以获得语音/声音信号。 从而可以用少量的计算和编码信息实现可伸缩编码。

    Speech coder and speech decoder
    25.
    发明申请

    公开(公告)号:US20060080091A1

    公开(公告)日:2006-04-13

    申请号:US11281386

    申请日:2005-11-18

    IPC分类号: G10L19/12

    摘要: A code excited linear prediction speech decoder is provided. An adaptive codebook generates an adaptive code vector. A random codebook generates a random code vector. A synthesis filter receives a signal based on the adaptive code vector and the random code vector, and performs linear prediction coefficient synthesis on the signal. The random codebook includes a pulse vector provider that provides a pulse vector having a signed unit pulse, a comparator that compares a value of adaptive codebook gain with a preset threshold value, a selector that selects a dispersion pattern from a plurality of dispersion patterns stored in a memory in accordance with a result of the comparison, and a generator that generates the dispersed vector by convoluting the pulse vector and the selected dispersion pattern.

    Excitation vector generator, speech coder and speech decoder
    27.
    发明申请
    Excitation vector generator, speech coder and speech decoder 有权
    激励矢量发生器,语音编码器和语音解码器

    公开(公告)号:US20050203736A1

    公开(公告)日:2005-09-15

    申请号:US11126171

    申请日:2005-05-11

    摘要: A noise canceller removes a noise component from an input speech signal. The noise canceller includes a noise cancellation coefficient adjuster that adjusts a noise cancellation coefficient to determine an amount of noise cancellation. A noise spectrum storage device stores an estimated noise spectrum. A noise estimator estimates a noise spectrum by comparing an input spectrum with a noise spectrum stored in the noise spectrum storage device. A noise canceling/spectrum compensator subtracts the noise spectrum stored in the noise spectrum storage device from the input spectrum based on a coefficient acquired by the noise cancellation coefficient adjuster.

    摘要翻译: 噪声消除器从输入语音信号中去除噪声分量。 噪声消除器包括噪声消除系数调节器,其调整噪声消除系数以确定噪声消除量。 噪声频谱存储装置存储估计的噪声谱。 噪声估计器通过将输入频谱与存储在噪声频谱存储装置中的噪声谱进行比较来估计噪声谱。 噪声消除/频谱补偿器基于由噪声消除系数调整器获取的系数从输入频谱中减去存储在噪声频谱存储装置中的噪声谱。

    Excitation vector generator, speech coder and speech decoder
    28.
    发明授权
    Excitation vector generator, speech coder and speech decoder 有权
    激励矢量发生器,语音编码器和语音解码器

    公开(公告)号:US06757650B2

    公开(公告)日:2004-06-29

    申请号:US09855708

    申请日:2001-05-16

    IPC分类号: G10L1904

    摘要: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

    摘要翻译: 传统CELP型语音编码器/解码器的随机码矢量读取部分和随机码本分别被替换为根据输入种子的值输出不同矢量流的振荡器和用于存储多个种子的种子存储部分。 这使得不必像固定码本(ROM)一样存储固定向量,从而显着地降低了存储器容量。

    Speech coding apparatus, linear prediction coefficient analyzing apparatus and noise reducing apparatus
    29.
    发明授权
    Speech coding apparatus, linear prediction coefficient analyzing apparatus and noise reducing apparatus 有权
    语音编码装置,线性预测系数分析装置和降噪装置

    公开(公告)号:US06205421B1

    公开(公告)日:2001-03-20

    申请号:US09475249

    申请日:1999-12-30

    申请人: Toshiyuki Morii

    发明人: Toshiyuki Morii

    IPC分类号: G10L2102

    摘要: A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained. Accordingly, because an appropriate coding module can be easily selected from a plurality of coding modules according to the coding module selecting rule, any allophone occurring in a reproduced speech can be prevented at a low calculation volume.

    摘要翻译: 通过语音分析单元分析样本语音以获得样本特征参数,并且根据多个编码模块中的每一个中的样本特征参数计算编码失真。 样本特征参数和编码失真由统计处理单元统计处理,以获得编码模块选择规则。 此后,当通过语音分析单元分析语音以获得特征参数时,根据编码模块选择规则,由编码模块选择单元从编码模块选择单元选择适当的编码模块,条件是特征参数的编码失真 在适当的编码模块中被最小化。 此后,将语音的特征参数编码在适当的编码模块中,并获得编码语音。 当编码语音被解码时,获得再现语音。 因此,由于可以根据编码模块选择规则从多个编码模块中容易地选择适当的编码模块,所以可以以较低的计算量来防止出现在再现语音中的任何异音素。

    Linear prediction coefficient analyzing apparatus for the
auto-correlation function of a digital speech signal
    30.
    发明授权
    Linear prediction coefficient analyzing apparatus for the auto-correlation function of a digital speech signal 有权
    用于数字语音信号的自相关函数的线性预测系数分析装置

    公开(公告)号:US6167373A

    公开(公告)日:2000-12-26

    申请号:US475248

    申请日:1999-12-30

    申请人: Toshiyuki Morii

    发明人: Toshiyuki Morii

    摘要: A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained. Accordingly, because an appropriate coding module can be easily selected from a plurality of coding modules according to the coding module selecting rule, any allophone occurring in a reproduced speech can be prevented at a low calculation volume.

    摘要翻译: 通过语音分析单元分析样本语音以获得样本特征参数,并且根据多个编码模块中的每一个中的样本特征参数计算编码失真。 样本特征参数和编码失真由统计处理单元进行统计处理,以获得编码模块选择规则。 此后,当通过语音分析单元分析语音以获得特征参数时,根据编码模块选择规则,由编码模块选择单元从编码模块选择单元选择适当的编码模块,条件是特征参数的编码失真 在适当的编码模块中被最小化。 此后,将语音的特征参数编码在适当的编码模块中,并获得编码语音。 当编码语音被解码时,获得再现语音。 因此,由于可以根据编码模块选择规则从多个编码模块中容易地选择适当的编码模块,所以可以以较低的计算量来防止出现在再现语音中的任何异音素。