Device and method for quantizing and inverse quantizing LPC filters in a super-frame
    21.
    发明授权
    Device and method for quantizing and inverse quantizing LPC filters in a super-frame 有权
    在超帧中量化和逆量化LPC滤波器的装置和方法

    公开(公告)号:US08712764B2

    公开(公告)日:2014-04-29

    申请号:US12501197

    申请日:2009-07-10

    IPC分类号: G10L19/04

    摘要: A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter.

    摘要翻译: 一种用于在包括帧序列的超帧中量化在序列的帧期间计算的LPC滤波器的装置和方法。 LPC滤波器量化装置和方法包括:绝对量化器,用于使用绝对量化首先量化LPC滤波器之一; 以及使用从相对于LPC滤波器中的至少一个先前量化的滤波器的绝对量化和差分量化组成的组中选择的量化模式的其它LPC滤波器的至少一个量化器。 对于逆量化,至少接收第一量化LPC滤波器,并且逆量化器使用绝对反量化对第一量化LPC滤波器进行逆量化。 如果接收到除了第一量化LPC滤波器之外的任何量化LPC滤波器,则逆量化器相对于至少一个先前接收的量化LPC滤波器,使用绝对反量子化和差分逆量化之一对该量化的LPC滤波器进行逆量化。

    Simultaneous time-domain and frequency-domain noise shaping for TDAC transforms
    22.
    发明授权
    Simultaneous time-domain and frequency-domain noise shaping for TDAC transforms 有权
    TDAC变换的同时时域和频域噪声整形

    公开(公告)号:US08626517B2

    公开(公告)日:2014-01-07

    申请号:US12905750

    申请日:2010-10-15

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/08 G10L19/12

    摘要: A frequency-domain noise shaping method and device interpolates a spectral shape and a time-domain envelope of a quantization noise in a windowed and transform-coded audio signal. In the method and device, transform coefficients of the windowed and transform-coded audio signal are split into a plurality of spectral bands. For each spectral band, a first gain representing a spectral shape of the quantization noise at a first transition between a first time window and a second time window is calculated, a second gain representing a spectral shape of the quantization noise at a second transition between the second time window and a third time window is calculated, and the transform coefficients of the second time window are filtered based on the first and second gains, to interpolate between the first and second transitions the spectral shape and the time-domain envelope of the quantization noise.

    摘要翻译: 频域噪声整形方法和装置在窗口化和变换编码的音频信号中内插量化噪声的频谱形状和时域包络。 在该方法和装置中,窗口化和变换编码的音频信号的变换系数被分成多个频谱带。 对于每个光谱带,计算表示在第一时间窗口和第二时间窗口之间的第一转换处的量化噪声的光谱形状的第一增益,第二增益表示在第一时间窗口和第二时间窗口之间的第二过渡处的量化噪声的光谱形状 计算第二时间窗口和第三时间窗口,并且基于第一和第二增益对第二时间窗口的变换系数进行滤波,以在第一和第二转换之间插值量化的频谱形状和时域包络 噪声。

    Flexible and Scalable Combined Innovation Codebook for Use in CELP Coder and Decoder
    23.
    发明申请
    Flexible and Scalable Combined Innovation Codebook for Use in CELP Coder and Decoder 有权
    灵活可扩展的组合创新代码用于CELP编码器和解码器

    公开(公告)号:US20120089389A1

    公开(公告)日:2012-04-12

    申请号:US13083900

    申请日:2011-04-11

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/12

    摘要: In a CELP coder, a combined innovation codebook coding device comprises a pre-quantizer of a first, adaptive-codebook excitation residual, and a CELP innovation-codebook search module responsive to a second excitation residual produced from the first, adaptive-codebook excitation residual. In a CELP decoder, a combined innovation codebook comprises a de-quantizer of pre-quantized coding parameters into a first excitation contribution, and a CELP innovation-codebook structure responsive to CELP innovation-codebook parameters to produce a second excitation contribution.

    摘要翻译: 在CELP编码器中,组合创新码本编码装置包括第一自适应码本激励残差的预量化器和响应于从第一自适应码本激励残差产生的第二激励残差的CELP创新码本搜索模块 。 在CELP解码器中,组合创新码本包括预量化编码参数到第一激励贡献的去量化器,以及响应于CELP创新码本参数以产生第二激励贡献的CELP创新码本结构。

    Audio Encoder and Decoder for Encoding and Decoding Frames of a Sampled Audio Signal
    24.
    发明申请
    Audio Encoder and Decoder for Encoding and Decoding Frames of a Sampled Audio Signal 有权
    音频编码器和解码器,用于对采样音频信号的帧进行编码和解码

    公开(公告)号:US20110173011A1

    公开(公告)日:2011-07-14

    申请号:US13004475

    申请日:2011-01-11

    IPC分类号: G10L19/00

    CPC分类号: G10L19/0212 G10L19/04

    摘要: An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.

    摘要翻译: 一种音频编码器,适于编码采样的音频信号的帧以获得编码的帧,其中帧包括多个时域音频采样。 音频编码器包括用于基于音频样本的帧来确定关于合成滤波器和预测域帧的系数的信息的预测编码分析阶段。 音频编码器还包括时间混叠引入变压器,用于将重叠的预测域帧转换到频域以获得预测域帧频谱,其中时间混叠引入变压器适于以重要取样的方式变换重叠的预测域帧 。 此外,音频编码器包括冗余减少编码器,用于根据系数和编码的预测域帧频谱对预测域帧频谱进行编码以获得编码的帧。

    Simultaneous Time-Domain and Frequency-Domain Noise Shaping for TDAC Transforms
    25.
    发明申请
    Simultaneous Time-Domain and Frequency-Domain Noise Shaping for TDAC Transforms 有权
    用于TDAC变换的同步时域和频域噪声整形

    公开(公告)号:US20110145003A1

    公开(公告)日:2011-06-16

    申请号:US12905750

    申请日:2010-10-15

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/00

    摘要: A frequency-domain noise shaping method and device interpolates a spectral shape and a time-domain envelope of a quantization noise in a windowed and transform-coded audio signal. In the method and device, transform coefficients of the windowed and transform-coded audio signal are split into a plurality of spectral bands. For each spectral band, a first gain representing a spectral shape of the quantization noise at a first transition between a first time window and a second time window is calculated, a second gain representing a spectral shape of the quantization noise at a second transition between the second time window and a third time window is calculated, and the transform coefficients of the second time window are filtered based on the first and second gains, to interpolate between the first and second transitions the spectral shape and the time-domain envelope of the quantization noise.

    摘要翻译: 频域噪声整形方法和装置在窗口化和变换编码的音频信号中内插量化噪声的频谱形状和时域包络。 在该方法和装置中,窗口化和变换编码的音频信号的变换系数被分成多个频谱带。 对于每个光谱带,计算表示在第一时间窗口和第二时间窗口之间的第一转换处的量化噪声的光谱形状的第一增益,第二增益表示在第一时间窗口和第二时间窗口之间的第二过渡处的量化噪声的光谱形状 计算第二时间窗口和第三时间窗口,并且基于第一和第二增益对第二时间窗口的变换系数进行滤波,以在第一和第二转换之间插值量化的频谱形状和时域包络 噪声。

    Method and device for low bit rate speech coding
    26.
    发明申请
    Method and device for low bit rate speech coding 有权
    低比特率语音编码的方法和装置

    公开(公告)号:US20060106600A1

    公开(公告)日:2006-05-18

    申请号:US11265440

    申请日:2005-11-01

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/12

    CPC分类号: G10L19/12

    摘要: A method for coding speech or other generic signals includes dividing a speech signal into a plurality of frames, and dividing at least one of the plurality of frames into at least two subframe units. A search for a fixed codebook contribution and an adaptive codebook contribution for subframe units is conducted. At least one subframe unit is selected to be coded without the fixed codebook contribution. The encoder may iteratively arrange and encode subframes differently for the same frame, and select for transmission that arrangement that minimizes an error measure across the frame. Various embodiments are shown, as are embodied computer programs, a decoder, and a communication system.

    摘要翻译: 用于对语音或其他通用信号进行编码的方法包括将语音信号划分为多个帧,并将多个帧中的至少一个划分为至少两个子帧单元。 进行对子帧单元的固定码本贡献和自适应码本贡献的搜索。 选择至少一个子帧单元进行编码,而不需要固定的码本贡献。 编码器可以对相同的帧不同地迭代地布置和编码子帧,并且选择用于传输使整个帧上的误差测量最小化的布置。 示出了各种实施例,如具体的计算机程序,解码器和通信系统。

    Multi-reference LPC filter quantization and inverse quantization device and method
    27.
    发明授权
    Multi-reference LPC filter quantization and inverse quantization device and method 有权
    多参考LPC滤波器量化和逆量化设备及方法

    公开(公告)号:US08332213B2

    公开(公告)日:2012-12-11

    申请号:US12501188

    申请日:2009-07-10

    IPC分类号: G10L21/00

    摘要: A multi-reference quantization device and method for quantizing an input LPC filter, comprises a plurality of differential quantizers using respective, different references, and a selector of a reference amongst the different references of the differential quantizers using a reference selection criterion. The input LPC filter is differentially quantized by the differential quantizer using the selected reference. A device and method for inverse quantizing a multi-reference differentially quantized LPC filter extracted from a bitstream, comprises an extractor from the bitstream of information about a reference amongst a plurality of possible references used for quantizing the multi-reference differentially quantized LPC filter, and a differential inverse quantizer using the reference corresponding to the extracted reference information to inverse quantize the multi-reference differentially quantized LPC filter.

    摘要翻译: 用于量化输入LPC滤波器的多参考量化装置和方法包括使用各自不同参考的多个差分量化器和使用参考选择准则的差分量化器的不同参考中的参考选择器。 输入LPC滤波器通过使用所选参考的差分量化器进行差分量化。 一种用于对从比特流提取的多参考差分量化LPC滤波器进行逆量化的装置和方法,包括来自比特流的提取器,该信息涉及用于量化多参考差分量化LPC滤波器的多个可能参考中的参考,以及 使用对应于所提取的参考信息的参考来对多参考差分量化LPC滤波器进行逆量化的差分逆量化器。

    FORWARD TIME-DOMAIN ALIASING CANCELLATION USING LINEAR-PREDICTIVE FILTERING
    28.
    发明申请
    FORWARD TIME-DOMAIN ALIASING CANCELLATION USING LINEAR-PREDICTIVE FILTERING 有权
    使用线性预测过滤的前向时域消除取消

    公开(公告)号:US20120022880A1

    公开(公告)日:2012-01-26

    申请号:US13006168

    申请日:2011-01-13

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L21/04

    摘要: In a coder, a method for producing forward aliasing cancellation (FAC) parameters for cancelling time-domain aliasing caused to a coded audio signal in a first transform-coded frame by a transition between the first transform-coded frame using a first coding mode with overlapping window and a second frame using a second coding mode with non-overlapping window, comprising: calculating a FAC target representative of a difference between the audio signal of the first frame prior to coding and a synthesis of the coded audio signal of the first transform-coded frame; and weighting the FAC target to produce the FAC parameters. In a decoder, weighted forward aliasing cancellation (FAC) parameters are received and inverse weighted to produce a FAC synthesis. Upon synthesis of the coded audio signal in the first frame, the time-domain aliasing is cancelled from the audio signal synthesis using the FAC synthesis.

    摘要翻译: 在编码器中,一种用于产生前向混叠消除(FAC)参数的方法,用于通过使用第一编码模式的第一变换编码帧与第一编码模式之间的转换来消除在第一变换编码帧中对编码音频信号造成的时域混叠, 重叠窗口和使用具有非重叠窗口的第二编码模式的第二帧,包括:计算表示编码之前的第一帧的音频信号与第一变换的编码音频信号的合成之间的差的FAC目标 编码框架 并加权FAC目标以产生FAC参数。 在解码器中,接收加权前向混叠消除(FAC)参数并进行逆加权以产生FAC合成。 在合成第一帧中的编码音频信号时,使用FAC合成从音频信号合成中消除时域混叠。

    Forward Time-Domain Aliasing Cancellation with Application in Weighted or Original Signal Domain
    29.
    发明申请
    Forward Time-Domain Aliasing Cancellation with Application in Weighted or Original Signal Domain 有权
    在加权或原始信号域中应用的前向时域混叠取消

    公开(公告)号:US20110153333A1

    公开(公告)日:2011-06-23

    申请号:US12821936

    申请日:2010-06-23

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/00

    摘要: The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects.

    摘要翻译: 本发明涉及从编码器向解码器发送的编码信号中的前向时域混叠消除的方法和装置。 在编码信号中计算与编码信号中的时域混叠相关的信息,并将其加到从编码器发送到解码器的比特流中。 解码器响应于包含在比特流中的信息,接收比特流并消除编码信号中的时域混叠。 该信息可以代表在第一编码模式下要编码的音频信号的帧与来自包括时域混叠效应的帧的解码信号之间的差异。

    Methods and devices for low-frequency emphasis during audio compression based on ACELP/TCX
    30.
    发明授权
    Methods and devices for low-frequency emphasis during audio compression based on ACELP/TCX 有权
    基于ACELP / TCX的音频压缩中低频重点的方法和设备

    公开(公告)号:US07933769B2

    公开(公告)日:2011-04-26

    申请号:US11708097

    申请日:2007-02-15

    申请人: Bruno Bessette

    发明人: Bruno Bessette

    IPC分类号: G10L19/04

    摘要: In a method and device for low-frequency emphasis, where the spectrum of a sound signal is transformed in a frequency domain and comprises transform coefficients grouped in a number of blocks, a maximum energy for one block having a position index is calculated. Also, a factor having a position index smaller than the position index of the block with maximum energy is calculated for each block. For each block, an energy of the block is calculated, the factor is computed from the calculated maximum energy and the computed energy of the block, and a gain is determined from the factor and applied to the transform coefficients of the block.

    摘要翻译: 在用于低频强调的方法和装置中,其中声频信号的频谱在频域中变换并且包括分组在多个块中的变换系数,计算具有位置索引的一个块的最大能量。 此外,针对每个块计算具有小于具有最大能量的块的位置索引的位置索引的因子。 对于每个块,计算块的能量,根据计算的最大能量和块的计算能量计算因子,并且根据因子确定增益并将其应用于块的变换系数。