公开(公告)号：WO2000011652A1

公开(公告)日：2000-03-02

申请号：PCT/US1999019134

申请日：1999-08-24

Applicant: CONEXANT SYSTEMS, INC.

Inventor： CONEXANT SYSTEMS, INC. ,  GAO, Yang

IPC: G10L19/08

CPC classification number: G10L19/125 ,  G10L19/002 ,  G10L19/005 ,  G10L19/012 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/18 ,  G10L19/265 ,  G10L21/0364 ,  G10L2019/0005 ,  G10L2019/0007 ,  G10L2019/0011

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech encoder also utilizes an adaptive weighting factor in the selection of a current pitch lag value from a plurality of pitch lag candidates. For example, if the speech encoder identifies an integer multiple timing relationship between any two pitch lag candidates, the pitch lag candidate with the smallest timing value is favored through adjustment of the weighting factor. Similarly, if a pitch lag candidate exhibits timing that corresponds to that of previous pitch lag values, the weighting factor is adjusted to favor that candidate.

Abstract translation: 多速率语音编解码器通过自适应地选择编码比特率模式来匹配通信信道限制，支持多种编码比特率模式。 在较高的比特率编码模式中，通过CELP（码激励线性预测）和其他相关联的建模参数准确地表达语音以产生更高质量的解码和再现。 为了在低比特率编码模式下实现高质量，语音编码器脱离了常规CELP编码器的严格波形匹配标准，并努力识别输入信号的显着感知特征。 为了支持较低比特率编码模式，应用了许多技术，其中许多技术涉及输入信号的分类。 对于所选择的每个比特率模式，选择多个固定或创新子码本来用于产生创新向量。 语音编码器还在从多个音调滞后候选中选择当前音调滞后值的同时利用自适应加权因子。 例如，如果语音编码器识别任何两个音调滞后候选之间的整数多个定时关系，则通过调整加权因子有利于具有最小定时值的音调滞后候选。 类似地，如果音调滞后候选表现出与先前的音调滞后值相对应的定时，则调整加权因子以有利于该候选。

公开(公告)号：WO2003003348A1

公开(公告)日：2003-01-09

申请号：PCT/US2002/011928

申请日：2002-04-16

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang ,  SU, Huan-Yu

IPC: G10L19/14

CPC classification number: G10L19/265 ,  G10L19/18 ,  G10L21/0364

Abstract: In a coding procedure, a spectral content of a speech signal is estimated. A preferential coding algorithm or preferential value of at least one coding parameter is selected based on the estimated spectral content of the speech signal. The speech signal is coded in accordance with the selected coding algorithm or the selected coding parameter to control the operation of one or more of the following: a pre-processing filter, a post-processing filter, a coding control coefficient, a weighting filter, a synthesis filter, and a quantization table.

Abstract translation: 在编码过程中，估计语音信号的频谱内容。 基于所估计的语音信号的频谱内容来选择优选编码算法或至少一个编码参数的优先值。 语音信号根据所选择的编码算法或选择的编码参数进行编码，以控制以下一个或多个的操作：预处理滤波器，后处理滤波器，编码控制系数，加权滤波器， 合成滤波器和量化表。

公开(公告)号：WO2000011658A1

公开(公告)日：2000-03-02

申请号：PCT/US1999019133

申请日：1999-08-24

Applicant: CONEXANT SYSTEMS, INC.

Inventor： CONEXANT SYSTEMS, INC. ,  SU, Huan-yu ,  GAO, Yang

IPC: G10L19/14

CPC classification number: G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/08 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/125 ,  G10L19/18 ,  G10L19/265 ,  G10L21/0364 ,  G10L2019/0005 ,  G10L2019/0007 ,  G10L2019/0011

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The fixed codebook contains pulse subcodebooks and noise-like subcodebooks. To assist in selection of one of the subcodebooks, an adaptive weighting approach is applied in a searching procedure wherein residual classification and various parameters are used to generate a weighting function that is used to favor one subcodebook over another. The pulse subcodebooks are favored to code pulse-like residuals, while the noise-like subcodebooks are favored to code noise-like residuals. The classification may involve identification of noise-like residuals, while the various parameters may comprise pitch correlation, signal to noise ratio, and average to peak ratio. Favoring involves an adjustment to a weighting factor applied to the subcodebooks.

Abstract translation: 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中，通过CELP（码激励线性预测）和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 为了支持较低比特率编码模式，应用了许多技术，其中许多技术涉及输入信号的分类。 对于所选择的每个比特率模式，选择多个固定或创新子码本来用于产生创新向量。 固定码本包含脉冲子码本和类噪声子码。 为了帮助选择一个子码本，在搜索过程中应用自适应加权方法，其中使用残差分类和各种参数来产生用于有利于一个子码本的加权函数。 脉冲子码本有利于编码类似脉冲的残差，而类似噪声的子代码则有利于编码类似噪声的残差。 该分类可以涉及噪声类残差的识别，而各种参数可以包括音调相关性，信噪比和平均峰值比。 喜好涉及到适用于子码本的加权因子的调整。

公开(公告)号：WO2003098598A1

公开(公告)日：2003-11-27

申请号：PCT/US2003/006335

申请日：2003-02-26

Applicant: CONEXANT SYSTEMS, INC.

Inventor： BENYASSINE, Adil ,  SHLOMOT, Eyal ,  SU, Huan-Yu ,  THYSSEN, Jes ,  GAO, Yang

IPC: G10L11/04

CPC classification number: H04W88/181 ,  G10L19/173

Abstract: There is provided transcoding of speech in a packet network environment. A decoder configured to receive a first bit-stream encoded according to a first coding scheme. The decoder decodes the bit-stream according to the first coding scheme, generates a plurality of first speech samples, and extracts a plurality of first speech parameters, which may include spectral characteristics, energy, pitch and/or pitch gain. A converter then converts the plurality first speech samples and plurality of first speech parameters to a plurality of second speech samples and a plurality of second speech parameters for use according to a second coding scheme. The first and second coding schemes may be, for example, G.711, G.723.1, G.726 or G.729, and may be parametric or non-parametric. An encoder receives the plurality of second speech samples and plurality of second speech parameters and generates a second bit-stream according to the second coding scheme.

Abstract translation: 在分组网络环境中提供语音转码。 解码器，被配置为接收根据第一编码方案编码的第一比特流。 解码器根据第一编码方案解码比特流，产生多个第一语音样本，并且提取多个第一语音参数，其可以包括频谱特性，能量，音调和/或音调增益。 然后，A转换器将多个第一语音样本和多个第一语音参数转换为多个第二语音样本和多个第二语音参数，以便根据第二编码方案使用。 第一和第二编码方案可以是例如G.711，G.723.1，G.726或G.729，并且可以是参数或非参数的。 编码器接收多个第二语音样本和多个第二语音参数，并根据第二编码方案生成第二比特流。

公开(公告)号：WO2003023764A1

公开(公告)日：2003-03-20

申请号：PCT/US2002/026817

申请日：2002-08-23

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang ,  SU, Huan-Yu

IPC: G10L13/00

CPC classification number: G10L19/265 ,  G10L21/0364

Abstract: A method for preparing a speech signal for encoding comprises determining whether the spectral content of an input speech signal is representative of a defined spectral characteristic (e.g., a defined characteristic slope). A frequency specific filter component of a weighting filter is controlled based on the determination of the spectral content of speech signal or/and its location in the encoder. A core weighting filter component of the weighting filter may be maintained regardless of the spectral content of the speech signal.

Abstract translation: 用于编制用于编码的语音信号的方法包括确定输入语音信号的频谱内容是否表示定义的频谱特性（例如，定义的特征斜率）。 加权滤波器的频率特定滤波器分量基于语音信号的频谱内容的确定和/或其在编码器中的位置而被控制。 无论语音信号的频谱内容如何，​​都可以维持加权滤波器的核心加权滤波器分量。

公开(公告)号：WO2002071396A1

公开(公告)日：2002-09-12

申请号：PCT/US2002/001846

申请日：2002-01-22

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang

IPC: G10L19/10

CPC classification number: G10L19/10 ,  G10L2019/0005

Abstract: A speech compression system with a fixed codebook structure and a search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A better way is used to calculate a criterion value, minimizing an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Abstract translation: 提出了一种具有固定码本结构和搜索程序的语音压缩系统，用于语音编码。 该系统能够将语音信号编码为比特流以用于后续解码以产生合成语音。 码本结构使用多个子码本。 每个子码本被设计成适合特定的一组语音信号。 使用更好的方法来计算标准值，最小化作为编码系统的一部分的最小化循环中的误差信号。 外部信号设置用于将编码语音传递到通信系统中的最大比特率速率。 语音压缩系统包括全速率编解码器，半速率编解码器，四分之一速率编解码器和八速率编解码器。 选择性地激活每个编解码器以以不同比特率对语音信号进行编码和解码，以在有限的平均比特率下提高合成语音的整体质量。

公开(公告)号：WO2002025634A3

公开(公告)日：2002-03-28

申请号：PCT/IB2001/001734

申请日：2001-09-17

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang ,  SU, Huan-Yu

IPC: G10L19/14

Abstract: A signal processing system is well suited for conditioning a speech signal prior to coding the speech signal to achieve enhanced perceptual quality of reproduced speech. The signal processing system may be incorporated into mobile or portable wireless communications devices, wireless infrastructure equipment, or both. The signal processing system includes a filtering arrangement for filtering an input speech signal to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

公开(公告)号：WO2002067247A1

公开(公告)日：2002-08-29

申请号：PCT/US2002/002984

申请日：2002-01-22

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang

IPC: G10L19/14

CPC classification number: G10L19/02 ,  G10L19/0204 ,  G10L19/0212

Abstract: Voiced speech preprocessing employs waveform interpolation or a harmonic model circuit to smooth a transition region and simplify speech coding. At low bit rates, the speech is coded by a system that maintains a high perceptual quality in the transition region from a voiced (quasi-periodic) portion of the speech signal to an unvoiced (non-periodic) portion of the speech signal. Similarly, the transition region from an unvoiced portion to a voiced portion is conditioned to maintain a high perceptual quality at a low bandwidth. The transition region from one type of voiced region to another type of voiced region is also smoothed. The transition region is smoothed to create a quasi-periodic speech signal.

Abstract translation: 语音预处理采用波形插值或谐波模型电路平滑过渡区域，简化语音编码。 在低比特率下，语音由系统编码，该系统在从语音信号的有声（准周期）部分到语音信号的无声（非周期性）部分的过渡区域中保持高感知质量。 类似地，从清音部分到有声部分的过渡区被调节以​​在低带宽下保持高感知质量。 从一种类型的浊音区域到另一种浊音区域的过渡区域也被平滑化。 平滑过渡区域以产生准周期性语音信号。

公开(公告)号：WO2000011653A1

公开(公告)日：2000-03-02

申请号：PCT/US1999019175

申请日：1999-08-24

Applicant: CONEXANT SYSTEMS, INC.

Inventor： CONEXANT SYSTEMS, INC. ,  GAO, Yang

IPC: G10L19/08

CPC classification number: G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/08 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/125 ,  G10L19/18 ,  G10L19/265 ,  G10L21/0364 ,  G10L2019/0005 ,  G10L2019/0007 ,  G10L2019/0011

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. The speech encoder continuously warps a weighted speech signal in long term preprocessing. The continuous warping is applied to a linear pitch lag contour that enables fast searching through linear time weighting. Optimal searching is performed within a limited range that is defined at least in part on sharpness and speech classification. The speech encoder generates the linear pitch lag contour from previous and current pitch lag values. Such continuous warping may also be applied in an open loop approach to the residual signal.

Abstract translation: 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中，通过CELP（码激励线性预测）和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 为了支持较低比特率编码模式，应用了许多技术，其中许多技术涉及输入信号的分类。 语音编码器在长期预处理中连续扭曲加权语音信号。 连续翘曲应用于能够通过线性时间加权进行快速搜索的线性音调滞后轮廓。 在至少部分地基于锐度和语音分类的有限范围内执行最佳搜索。 语音编码器从先前和当前音调滞后值产生线性音调滞后轮廓。 这种连续翘曲也可以以开环方式应用于残余信号。

公开(公告)号：WO2002054380A3

公开(公告)日：2002-07-11

申请号：PCT/US2001/046778

申请日：2001-12-10

Applicant: CONEXANT SYSTEMS, INC.

Inventor： GAO, Yang

IPC: G10L19/12

Abstract: Pulses representing the excitation signal (P1, P2, P3) are commonly represented as an impulse. High frequency noise will be added to each pulse in order to provide a better quality sound.