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公开(公告)号:US20190272838A1
公开(公告)日:2019-09-05
申请号:US16417195
申请日:2019-05-20
Applicant: Huawei Technologies Co., Ltd.
Abstract: A decoder for processing an audio signal receives an audio bitstream, decodes the bitstream to obtain a set of spectral frequency parameters that are arranged in an order of frequencies, determines a minimum spectral frequency parameter difference from a plurality of calculated spectral frequency parameter differences, determines a start frequency bin for predicting a high band excitation signal according to the minimum spectral frequency parameter difference, generates the high band excitation signal by selecting a frequency band with a preset bandwidth selected from a low band excitation signal according to the start frequency bin, and synthesizes a wideband signal based on the generated high band excitation signal.
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公开(公告)号:US20190251980A1
公开(公告)日:2019-08-15
申请号:US16396253
申请日:2019-04-26
Applicant: HUAWEI TECHNOLOGIES CO., LTD.
IPC: G10L19/083 , G10L19/02 , G10L21/038 , G10L25/93 , G10L19/005 , G10L21/0388
Abstract: The present disclosure relates methods and apparatus for recovering a lost frame in a received audio signal. One example method includes obtaining an initial high-frequency band signal of a current lost frame in the received audio signal, calculating a ratio R, where the ratio R is a ratio of a high frequency excitation energy of a previous frame of the current lost frame to a high frequency excitation energy of the current lost frame, obtaining a global gain of the current lost frame according to the ratio R and a global gain of the previous frame of the current lost frame, and recovering a high-frequency band signal of the current lost frame according to the initial high-frequency band signal of the current lost frame and the global gain of the current lost frame.
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公开(公告)号:US20140114670A1
公开(公告)日:2014-04-24
申请号:US14145632
申请日:2013-12-31
Applicant: Huawei Technologies Co., Ltd.
IPC: G10L19/008
CPC classification number: G10L19/265 , G10L19/008 , G10L19/0204 , G10L19/087 , G10L19/18 , G10L21/038 , G10L25/18
Abstract: Example embodiments described herein generally provide for adaptive audio signal coding of low-frequency and high-frequency audio signals. More specifically, audio signals are categorized into high-frequency audio signals and low-frequency audio signals. Then, based on a set coding and/or characteristics of the low-frequency audio signals, the low-frequency coding manner is selected. Similarly, but in addition to, a bandwidth extension mode to code the high-frequency audio signals is selected according to the low-frequency coding manner and/or characteristics of the audio signals.
Abstract translation: 本文描述的示例性实施例通常提供低频和高频音频信号的自适应音频信号编码。 更具体地,音频信号被分类为高频音频信号和低频音频信号。 然后,基于低频音频信号的集合编码和/或特性,选择低频编码方式。 类似地,除了根据低频编码方式和/或音频信号的特性来选择对高频音频信号进行编码的带宽扩展模式。
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公开(公告)号:US20230119826A1
公开(公告)日:2023-04-20
申请号:US18065043
申请日:2022-12-13
Applicant: HUAWEI TECHNOLOGIES CO., LTD.
Inventor: Eyal SHLOMOT , Haiting LI , Zexin LIU
IPC: G10L19/008 , G10L19/032
Abstract: Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determined based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal, a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.
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公开(公告)号:US20220386064A1
公开(公告)日:2022-12-01
申请号:US17879114
申请日:2022-08-02
Applicant: HUAWEI TECHNOLOGIES CO., LTD.
Inventor: Gavin KEARNEY , Cal ARMSTRONG , Bin WANG , Zexin LIU
Abstract: M audio signals are obtained by processing an audio signal by M virtual speakers; M first HRTFs and M second HRTFs are obtained, where the M first HRTFs corresponding to a left ear position, and the M second HRTFs corresponding to a right ear position; high-band impulse responses of some of the M first HRTFs are modified to obtain modified first target HRTFs, and high-band impulse responses of some of the M second HRTFs are modified to obtain modified second target HRTFs; a first target audio signal corresponding to the left ear position is obtained based on the modified first target HRTFs and un-modified first HRTFs, and the M audio signals; and a second target audio signal corresponding to the right ear position is obtained based on the modified second HRTFs, un-modified second target HRTFs, and the M audio signals.
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公开(公告)号:US20210118456A1
公开(公告)日:2021-04-22
申请号:US17136028
申请日:2020-12-29
Applicant: HUAWEI TECHNOLOGIES CO., LTD.
Inventor: Eyal SHLOMOT , Haiting LI , Zexin LIU
IPC: G10L19/008 , G10L19/032
Abstract: Various embodiments provide a method and an apparatus for determining a weighting factor during stereo signal encoding. In those embodiments, a parameter value corresponding to the encoding mode of the to-be-encoded signal is determining based on an encoding mode of a to-be-encoded signal in a stereo signal and a correspondence between an encoding mode and a parameter value. Based on the determined parameter value and an energy spectrum of a linear prediction filter corresponding to an original line spectral frequency parameter of the to-be-encoded signal is a weighting factor for calculating a distance between the original line spectral frequency parameter and a target original line spectral frequency parameter is calculated.
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公开(公告)号:US20170110137A1
公开(公告)日:2017-04-20
申请号:US15391339
申请日:2016-12-27
Applicant: HUAWEI TECHNOLOGIES CO., LTD.
CPC classification number: G10L19/12 , G10L19/0204 , G10L19/0208 , G10L19/26
Abstract: Embodiments of the present invention provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.
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公开(公告)号:US20160118054A1
公开(公告)日:2016-04-28
申请号:US14981956
申请日:2015-12-29
Applicant: HUAWEI TECHNOLOGIES CO.,LTD.
IPC: G10L19/005 , G10L21/0232 , G10L25/93 , G10L19/02
CPC classification number: G10L19/005 , G10L19/0208 , G10L21/0232 , G10L25/93 , G10L2025/937
Abstract: A method for recovering lost frame in a media bitstream is provided. When a frame loss event occurs, a decoder obtains a synthesized high frequency band signal of a current lost frame, and recovery information related to the current lost frame. The decoder determines a global gain gradient of the current lost frame, and further determines a global gain of the current lost frame according to the global gain gradient and a global gain of each frame in previous M frames of the current lost frame. A high frequency band signal of the current lost frame is obtained by adjusting the synthesized high frequency band signal of the current lost frame according to the global gain and a subframe gain of the current lost frame. The process enables natural and smooth transitions of the high frequency band signal between the frames, and attenuates noises in the high frequency band signal.
Abstract translation: 提供了一种用于恢复媒体比特流中的丢帧的方法。 当帧丢失事件发生时,解码器获得当前丢失帧的合成高频带信号,以及与当前丢帧相关的恢复信息。 解码器确定当前丢失帧的全局增益梯度,并且还根据当前丢失帧的先前M帧中的全局增益梯度和每帧的全局增益来确定当前丢失帧的全局增益。 通过根据当前丢失帧的全局增益和子帧增益调整当前丢失帧的合成高频带信号,获得当前丢帧的高频带信号。 该过程使帧之间的高频信号能够自然平滑地转换,并且衰减高频信号中的噪声。
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公开(公告)号:US20160111104A1
公开(公告)日:2016-04-21
申请号:US14984703
申请日:2015-12-30
Applicant: HUAWEI TECHNOLOGIES CO.,LTD.
IPC: G10L19/02 , G10L19/032 , G10L19/002
CPC classification number: G10L19/0204 , G10L19/002 , G10L19/032
Abstract: Embodiments of the present disclosure provide signal encoding and decoding methods and devices. The method includes: determining, a quantity k of subbands to be encoded, where i is a positive number, and k is a positive integer; selecting, according to quantized envelopes of all subbands, k subbands from all the subbands, or selecting k subbands from all subbands according to a psychoacoustic model; and performing a first-time encoding operation on spectral coefficients of the k subbands. In the embodiments of the present disclosure, the quantity k of subbands to be encoded is determined according to the quantity of available bits and the first saturation threshold, and encoding is performed on the k subbands that are selected from all the subbands, instead of on an entire frequency band, which can reduce spectrum holes of a signal obtained through decoding, and therefore, can improve auditory quality of an output signal.
Abstract translation: 本公开的实施例提供信号编码和解码方法和装置。 该方法包括:确定要编码的子带的数量k,其中i是正数,k是正整数; 根据所有子带的量化包络选择来自所有子带的k个子带,或者根据心理声学模型从所有子带中选择k个子带; 以及对k个子带的频谱系数执行第一次编码操作。 在本公开的实施例中,根据可用比特量和第一饱和阈值来确定要编码的子带的数量k,并且对从所有子带中选择的k个子带进行编码,而不是在 整个频带,其可以减少通过解码获得的信号的频谱孔,并且因此可以提高输出信号的听觉质量。
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公开(公告)号:US20150302860A1
公开(公告)日:2015-10-22
申请号:US14789755
申请日:2015-07-01
Applicant: Huawei Technologies Co., Ltd.
Inventor: Fengyan QI , Zexin LIU , Lei MIAO
IPC: G10L19/032 , G10L19/002
CPC classification number: G10L19/032 , G10L19/002 , G10L19/0204 , G10L19/028
Abstract: An audio signal encoding method is provided. The method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.
Abstract translation: 提供了一种音频信号编码方法。 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; 根据量化的子带归一化因子,或根据量化的子带归一化因子和比特率信息确定比特分配的信号带宽; 在所确定的信号带宽内分配用于子带的比特; 以及根据为每个子带分配的比特来对音频信号的频谱系数进行编码。 根据本发明的实施例,在编码和解码期间,根据量化的子带归一化因子和比特率信息来确定比特分配的信号带宽。 以这种方式,通过集中位来有效地对确定的信号带宽进行编码和解码,并提高音频质量。
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