公开(公告)号：US09123329B2

公开(公告)日：2015-09-01

申请号：US13708611

申请日：2012-12-07

Applicant: Huawei Technologies Co., Ltd.

Inventor： Lei Miao ,  Chen Hu ,  Wenhai Wu ,  Yue Lang

IPC: H04R5/00 ,  G10L19/008

CPC classification number: H04S3/00 ,  G10L19/00 ,  G10L19/008

Abstract: Embodiments of the present invention provide a method and an apparatus for generating a sideband residual signal. The method includes: comparing energy of a first signal input by a first sound channel with energy of a second signal input by a second sound channel; if the energy of the first signal is greater than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the first signal; and if the energy of the first signal is smaller than the energy of the second signal, generating a sideband residual signal by allocating a monophonic quantization error to the second signal. By using the method and apparatus provided in the embodiments of the present invention, it can be avoided that a monophonic quantization error has a greater impact on a signal whose energy is smaller.

Abstract translation: 本发明的实施例提供一种用于生成边带残留信号的方法和装置。 该方法包括：将通过第一声道输入的第一信号的能量与由第二声道输入的第二信号的能量进行比较; 如果第一信号的能量大于第二信号的能量，则通过向第一信号分配单声道量化误差来产生边带残差信号; 并且如果第一信号的能量小于第二信号的能量，则通过向第二信号分配单声道量化误差来产生边带残留信号。 通过使用本发明实施例中提供的方法和装置，可以避免单声道量化误差对能量较小的信号的影响较大。

公开(公告)号：US20150206541A1

公开(公告)日：2015-07-23

申请号：US14675031

申请日：2015-03-31

Applicant: Huawei Technologies Co., Ltd.

Inventor： Fengyan Qi ,  Zexin Liu ,  Lei Miao

IPC: G10L19/002

CPC classification number: G10L19/002 ,  G10L19/0204 ,  G10L19/032 ,  G10L19/035

Abstract: A method and an apparatus for allocating bits of an audio signal. The method includes dividing a frequency band of an audio signal into multiple sub-bands, and quantizing a sub-band normalization factor of each sub-band; classifying the multiple sub-bands into multiple groups, and acquiring a sum of intra-group sub-band normalization factors of each group; performing initial inter-group bit allocation to determine the initial number of bits of each group; performing secondary inter-group bit allocation to allocate coding bits of the audio signal to at least one group; and allocating the bits of the audio signal to sub-bands in the group. The present invention can, by means of grouping, ensure relatively stable allocation in a previous frame and a next frame and reduce an impact of global allocation on local discontinuity in a case of low and medium bit rates.

Abstract translation: 一种用于分配音频信号的位的方法和装置。 该方法包括将音频信号的频带划分成多个子带，并量化每个子带的子带归一化因子; 将多个子带分为多个组，并获取每组的组内子带归一化因子之和; 执行初始组间比特分配以确定每组的初始比特数; 执行次要组间比特分配以将音频信号的编码比特分配给至少一个组; 并将音频信号的比特分配给组中的子带。 本发明可以通过分组确保在先前帧和下一帧中的相对稳定的分配，并且在低和中比特率的情况下减少全局分配对局部不连续性的影响。

公开(公告)号：US20150095038A1

公开(公告)日：2015-04-02

申请号：US14562494

申请日：2014-12-05

Applicant: Huawei Technologies Co., Ltd.

Inventor： Chen Hu ,  Zexin Liu ,  Lei Miao

IPC: G10L19/26

CPC classification number: G10L19/265 ,  G10L19/012 ,  G10L19/18 ,  G10L19/22

Abstract: The present disclosure provides a speech/audio signal processing method based on wideband switching and a coding apparatus. The method includes: if a first wideband speech/audio signal is a harmonic signal, adjusting a determining condition for determining that a second wideband speech/audio signal is a harmonic signal, to obtain a first determining condition, where the first wideband speech/audio signal is a signal before wideband switching, and the second wideband speech/audio signal is a signal after the wideband switching; and determining, according to the first determining condition, whether the second wideband speech/audio signal is a harmonic signal. In the case of wideband switching, signal types of speech/audio signals remain as consistent as possible before and after the switching, so that continuity of the speech/audio signal decoded by a decoder device is ensured as much as possible, further improving speech communication service quality.

Abstract translation: 本公开提供了一种基于宽带交换和编码装置的语音/音频信号处理方法。 该方法包括：如果第一宽带语音/音频信号是谐波信号，则调整用于确定第二宽带语音/音频信号是谐波信号的确定条件，以获得第一确定条件，其中第一宽带语音/音频 信号是宽带切换前的信号，第二宽带语音/音频信号是宽带切换后的信号; 以及根据所述第一确定条件确定所述第二宽带语音/音频信号是否是谐波信号。 在宽带切换的情况下，语音/音频信号的信号类型在切换之前和之后保持一致，从而尽可能地确保由解码器设备解码的语音/音频信号的连续性，进一步改善语音通信 服务质量。

公开(公告)号：US08712763B2

公开(公告)日：2014-04-29

申请号：US13943812

申请日：2013-07-17

Applicant: Huawei Technologies Co., Ltd.

Inventor： Dejun Zhang ,  Lei Miao ,  Jianfeng Xu ,  Fengyan Qi ,  Qing Zhang ,  Lixiong Li ,  Fuwei Ma ,  Yang Gao

IPC: G10L21/00

CPC classification number: G10L19/09 ,  G10L19/0017 ,  H04N19/50

Abstract: The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag value according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag value is a first value; obtaining a second-domain predictive signal according to the LP processing result when the long-term flag value is a second value; converting the second-domain predictive signal into a first-domain predictive signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract translation: 本公开涉及用于对信号进行编码和解码的方法，装置和系统。 编码方法包括：将第一域信号转换为第二域信号; 对第二域信号执行线性预测（LP）处理和长期预测（LTP）处理; 根据决策标准获得长期标志值; 当长期标志值为第一值时，根据LP处理结果和LTP处理结果获得第二域预测信号; 当长期标志值是第二值时，根据LP处理结果获得第二域预测信号; 将所述第二域预测信号转换为第一域预测信号，以及计算第一域预测残差信号; 并输出包括第一域预测残差信号的比特流。

公开(公告)号：US20140006036A1

公开(公告)日：2014-01-02

申请号：US13956093

申请日：2013-07-31

Applicant: Huawei Technologies Co., Ltd.

Inventor： Fengyan Qi ,  Lei Miao ,  Qing Zhang

IPC: G10L19/00

CPC classification number: G10L19/0019 ,  H03M7/30

Abstract: A coding method includes: obtaining a value of each sample of an input data frame; determining pulse samples and non-pulse samples in the input data frame according to the distribution of values of samples of the input data frame; encoding the determined pulse samples in the input data frame in a first coding mode to obtain a first data stream; encoding the determined non-pulse samples in the input data frame in a second coding mode to obtain a second data stream; and multiplexing the first data stream and the second data stream to obtain an output coded data stream of the input data frame. The technical solution under the present disclosure reduces the number of bits required for encoding the entire data frame is reduced, and improves the compression efficiency of the data frame with a wide dynamic range.

Abstract translation: 编码方法包括：获取输入数据帧的每个样本的值; 根据输入数据帧的样本值的分布，确定输入数据帧中的脉冲样本和非脉冲样本; 以第一编码模式对输入数据帧中确定的脉冲样本进行编码以获得第一数据流; 以第二编码模式对输入数据帧中确定的非脉冲样本进行编码，以获得第二数据流; 并且多路复用第一数据流和第二数据流，以获得输入数据帧的输出编码数据流。 本公开的技术方案减少了整个数据帧的编码所需的位数减少，并提高了宽动态范围的数据帧的压缩效率。

公开(公告)号：US20130236022A1

公开(公告)日：2013-09-12

申请号：US13850655

申请日：2013-03-26

Applicant: HUAWEI TECHNOLOGIES CO., LTD.

Inventor： David Virette ,  Yue Lang ,  Lei Miao ,  Wenhai Wu

IPC: G10L19/008

CPC classification number: G10L19/008 ,  G10L19/025 ,  G10L21/0364

Abstract: According to the invention, a device (101, 101′) for postprocessing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver (103; 103′) for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal and a classification indication indicating a transient type of the at least one channel signal, wherein the classification indication is associated to the at least one channel signal, and a postprocessor (105; 105′) for postprocessing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication.

Abstract translation: 根据本发明，描述了用于后处理多信道信号的多个信道信号的至少一个信道信号的装置（101,101'），所述至少一个信道信号是通过解码的下混合信号由 低比特率音频编码/解码系统，该装置包括：接收机（103; 103'），用于接收从解码缩混信号产生的至少一个信道信号，解码缩混信号的时间包络和分类指示 指示所述至少一个信道信号的瞬时类型，其中所述分类指示与所述至少一个信道信号相关联;以及后处理器（105; 105'），用于基于所述至少一个信道信号的时间包络来后处理所述至少一个信道信号 解码的下混合信号由相应的加权因子加权并且依赖于分类指示。

公开(公告)号：US20130124216A1

公开(公告)日：2013-05-16

申请号：US13738786

申请日：2013-01-10

Applicant: Huawei Technologies Co., Ltd.

Inventor： Chen Hu ,  Zexin Liu ,  Lei Miao ,  Longyin Chen ,  Qing Zhang ,  Wei Xiao ,  Marcel Taddei Herve

IPC: G10L21/00

CPC classification number: G10L21/00 ,  G10L19/24

Abstract: A method, device, and system for signal encoding and decoding are disclosed. The method includes: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. According to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract translation: 公开了用于信号编码和解码的方法，装置和系统。 该方法包括：编码核心层信号以获得核心层信号码; 选择需要根据核心层信号码的增强层信号编码和增强层可以使用的位数的增强采样点; 获得增强采样点的增强层信号码; 并输出比特流，其中比特流包括核心层信号码和增强层信号码。 根据增强层可以使用的比特数，选择需要增强层信号编码的增强采样点; 所选择的增强采样点的增强层信号被编码和解码; 当没有足够的比特可用于增强层时，可以提高核心层的增强质量。

公开(公告)号：US20250124932A1

公开(公告)日：2025-04-17

申请号：US18928904

申请日：2024-10-28

Applicant: Huawei Technologies Co., Ltd.

Inventor： Zexin Liu ,  Lei Miao

IPC: G10L19/008 ,  G10L19/02 ,  G10L19/032

Abstract: A multi-channel signal encoding method includes determining a downmixed signal of a first channel signal and a second channel signal, determining an initial reverberation gain parameter of the first channel signal and the second channel signal, determining a target reverberation gain parameter of the first channel signal and the second channel signal based on a correlation between the first channel signal and the downmixed signal, a correlation between the second channel signal and the downmixed signal, and the initial reverberation gain parameter, quantizing the first channel signal and the second channel signal based on the downmixed signal and the target reverberation gain parameter, and writing a quantized first channel signal and a quantized second channel signal into a bitstream.

公开(公告)号：US20250087219A1

公开(公告)日：2025-03-13

申请号：US18810463

申请日：2024-08-20

Applicant: HUAWEI TECHNOLOGIES CO., LTD.

Inventor： Zexin Liu ,  Fengyan Qi ,  Lei Miao

IPC: G10L19/002 ,  G10L19/005 ,  G10L19/02 ,  G10L19/028

Abstract: In a method to decode signals, a computing device decodes spectral coefficients of a current frame are grouped into a plurality of sub-bands. The computing device classifies a sub-band as a bit allocation unsaturated sub-band based on an average quantity of allocated bits per spectral coefficient of a sub-band of the plurality of sub-bands and a threshold. The computing device obtains a noise filling gain based on an envelope of the sub-band, and obtains a reconstructed spectral coefficient of the sub-band by performing noise filling based on the noise filling gain. The computing device then obtains a frequency domain audio signal based on spectral coefficients in the sub-band obtained by decoding and the reconstructed spectral coefficient.

公开(公告)号：US12230283B2

公开(公告)日：2025-02-18

申请号：US18449281

申请日：2023-08-14

Applicant: Huawei Technologies Co., Ltd.

Inventor： Eyal Shlomot ,  Haiting Li ,  Lei Miao

IPC: G10L19/008 ,  H04S1/00

Abstract: A stereo signal processing method includes performing delay estimation on a stereo signal of a current frame to determine an inter-channel time difference of the current frame, identifying a sign of the inter-channel time difference of the current frame is different from a sign of an inter-channel time difference of a previous frame of the current frame, performing delay alignment processing on the first-channel signal of the current frame based on the inter-channel time difference of the current frame, and performing delay alignment processing on the second-channel signal of the current frame based on the inter-channel time difference of the previous frame.