公开(公告)号：US08712764B2

公开(公告)日：2014-04-29

申请号：US12501197

申请日：2009-07-10

申请人： Philippe Gournay ,  Bruno Bessette ,  Redwan Salami

发明人： Philippe Gournay ,  Bruno Bessette ,  Redwan Salami

IPC分类号： G10L19/04

CPC分类号： G10L19/06 ,  G10L19/18 ,  H03M7/3082

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。

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

CPC分类号： G10L19/26 ,  G10L19/0204 ,  G10L19/0212 ,  G10L19/032 ,  G10L19/18 ,  G10L21/0208 ,  G10L2019/0008

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。

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

CPC分类号： G10L19/12 ,  G10L19/0212 ,  G10L19/09 ,  G10L19/24

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。

公开(公告)号：US08036885B2

公开(公告)日：2011-10-11

申请号：US12620394

申请日：2009-11-17

申请人： Bruno Bessette ,  Redwan Salami ,  Roch Lefebvre

发明人： Bruno Bessette ,  Redwan Salami ,  Roch Lefebvre

IPC分类号： G10L11/04 ,  G10L19/00 ,  G10L19/12

CPC分类号： G10L19/26 ,  G10L2019/0011

摘要： A pitch search method and device for digitally encoding a wideband signal, in particular but not exclusively a speech signal, in view of transmitting, or storing, and synthesizing this wideband sound signal. The new method and device which achieve efficient modeling of the harmonic structure of the speech spectrum uses several forms of low pass filters applied to a pitch codevector, the one yielding higher prediction gain (i.e. the lowest pitch prediction error) is selected and the associated pitch codebook parameters are forwarded.

摘要翻译： 考虑到发送或存储和合成该宽带声音信号，用于对宽带信号进行数字编码的音调搜索方法和装置，特别地但不排他地是语音信号。 实现语音频谱谐波结构的有效建模的新方法和装置使用几种形式的应用于音调码矢量的低通滤波器，选择产生较高预测增益（即，最低音调预测误差）的一种形式，并且相关音调 码本参数被转发。

公开(公告)号：US20110173011A1

公开(公告)日：2011-07-14

申请号：US13004475

申请日：2011-01-11

申请人： Ralf Geiger ,  Bernhard Grill ,  Bruno Bessette ,  Philippe Gournay ,  Guillaume Fuchs ,  Markus Multrus ,  Max Neuendorf ,  Gerald Schuller

发明人： Ralf Geiger ,  Bernhard Grill ,  Bruno Bessette ,  Philippe Gournay ,  Guillaume Fuchs ,  Markus Multrus ,  Max Neuendorf ,  Gerald Schuller

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。

公开(公告)号：US20110173004A1

公开(公告)日：2011-07-14

申请号：US12664010

申请日：2007-12-28

申请人： Bruno Bessette ,  Jimmy Lapierre ,  Vladimir Malenovsky ,  Roch Lefebvre ,  Redwan Salami

发明人： Bruno Bessette ,  Jimmy Lapierre ,  Vladimir Malenovsky ,  Roch Lefebvre ,  Redwan Salami

IPC分类号： G10L19/00

CPC分类号： G10L19/26 ,  G10L19/005 ,  G10L19/24 ,  G10L25/93

摘要： A device and method for shaping noise during encoding of an input sound signal comprise pre-emphasizing the input signal or a decoded signal from a given sound signal codec to produce a pre-emphasized signal, computing a filter transfer function based on the pre-emphasized signal, and shaping the noise by filtering the noise through the transfer function to produce a shaped noise signal, wherein the noise shaping comprises producing a noise feedback. A device and method for noise shaping in a multilayer codec, including at least Layer 1 and 2, comprise: at an encoder, producing an encoded sound signal in Layer 1 including Layer 1 noise shaping, and producing a Layer 2 enhancement signal; at a decoder, decoding the Layer 1 encoded sound signal to produce a synthesis signal, decoding the enhancement signal, computing a filter transfer function based on the synthesis signal, filtering the enhancement signal through the transfer function to produce a Layer 2 filtered enhancement signal, and adding the filtered enhancement signal to the synthesis signal to produce an output signal including contributions from Layer 1 and 2.

摘要翻译： 用于在编码输入声音信号期间整形噪声的装置和方法包括预先强调来自给定声音信号编解码器的输入信号或解码信号以产生预加重信号，基于预先强调的信号计算滤波器传递函数 信号和整形噪声，通过传递函数对噪声进行滤波以产生成形噪声信号，其中噪声整形包括产生噪声反馈。 包括至少第1层和第2层的多层编解码器中的噪声整形的装置和方法包括：在编码器处，产生包括层1噪声整形的层1中的编码声音信号，并产生第2层增强信号; 在解码器处，解码第1层编码声音信号以产生合成信号，对增强信号进行解码，基于合成信号计算滤波器传递函数，通过传递函数对增强信号进行滤波，以产生第2层滤波的增强信号， 并将经滤波的增强信号加到合成信号上以产生包括来自层1和2的贡献的输出信号。

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

CPC分类号： G10L19/26 ,  G10L19/0204 ,  G10L19/0212 ,  G10L19/032 ,  G10L19/18 ,  G10L21/0208 ,  G10L2019/0008

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。

公开(公告)号：US07529660B2

公开(公告)日：2009-05-05

申请号：US10515553

申请日：2003-05-30

申请人： Bruno Bessette ,  Claude Laflamme ,  Milan Jelinek ,  Roch Lefebvre

发明人： Bruno Bessette ,  Claude Laflamme ,  Milan Jelinek ,  Roch Lefebvre

IPC分类号： G10L19/02 ,  G10L21/02

CPC分类号： G10L21/0364 ,  G10L19/26 ,  G10L21/0232

摘要： In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

摘要翻译： 考虑到提高该解码声音信号的感知质量，对解码声音信号进行后处理的方法和装置中，解码声音信号被分成多个频率子带信号，后处理应用于 至少一个频率子带信号。 在对该至少一个频率子带信号进行后处理之后，可以添加频率子带信号以产生输出的后处理解码声音信号。 以这种方式，后处理可以被定位到期望的子带或子带，而使其他子带几乎不变。

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。

公开(公告)号：US08930198B2

公开(公告)日：2015-01-06

申请号：US13004385

申请日：2011-01-11

申请人： Bernhard Grill ,  Roch Lefebvre ,  Bruno Bessette ,  Jimmy Lapierre ,  Philippe Gournay ,  Redwan Salami ,  Stefan Bayer ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Ralf Geiger ,  Johannes Hilpert ,  Ulrich Kraemer ,  Jeremie Lecomte ,  Markus Multrus ,  Max Neuendorf ,  Harald Popp ,  Nikolaus Rettelbach

发明人： Bernhard Grill ,  Roch Lefebvre ,  Bruno Bessette ,  Jimmy Lapierre ,  Philippe Gournay ,  Redwan Salami ,  Stefan Bayer ,  Guillaume Fuchs ,  Stefan Geyersberger ,  Ralf Geiger ,  Johannes Hilpert ,  Ulrich Kraemer ,  Jeremie Lecomte ,  Markus Multrus ,  Max Neuendorf ,  Harald Popp ,  Nikolaus Rettelbach

IPC分类号： G10L19/00 ,  G10L19/18 ,  G10L19/16 ,  G10L19/008 ,  G10L19/02

CPC分类号： G10L19/008 ,  G10L19/0017 ,  G10L19/0212 ,  G10L19/173 ,  G10L19/18 ,  G10L2019/0008

摘要： An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

摘要翻译： 音频编码器具有第一信息汇集编码分支，第二信息源或SNR定向编码分支，以及用于在第一编码分支和第二编码分支之间进行切换的开关，其中第二编码分支具有转换器到特定的域 并且其中第二编码分支还具有特定的域编码分支，以及特定频域编码分支，以及用于在特定的域编码分支与特定频域编码分支之间切换的附加开关。 音频解码器具有第一域解码器，用于对信号进行解码的第二域解码器，以及用于在解码器之间切换的第三域解码器和两个级联开关。