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    • 11. 发明申请
    • Digital signal encoding method and apparatus using plural lookup tables
    • 使用多个查找表的数字信号编码方法和装置
    • US20050254588A1
    • 2005-11-17
    • US11080409
    • 2005-03-16
    • Dohyung KimJunghoe KimYangseock SeoShihwa LeeSangwook Kim
    • Dohyung KimJunghoe KimYangseock SeoShihwa LeeSangwook Kim
    • G10L19/00G10L19/02H03M7/30H04K1/10
    • G10L19/035
    • A digital signal encoding method and apparatus using a plurality of lookup tables. The method includes: preparing a plurality of lookup tables storing numbers of allocated bits for encoding frequency bands of an input signal according to a characteristic of the input signal in a predetermined number of addresses; dividing an input signal in the time domain into signals in predetermined frequency bands; calculating address values of the frequency bands; selecting one of the plurality of lookup tables according to the characteristic of the input signal; extracting numbers of allocated bits of addresses having the calculated address values from the selected lookup table with respect to the frequency bands and allocating the numbers of bits to the frequency bands; and generating a bitstream by quantizing the input signal according to the numbers of allocated bits. Bit amount control suitable for a characteristic of an input signal can be performed by extracting numbers of allocated bits of frequency bands from an optimal lookup table selected according to the characteristic of the input signal. Also, an additional computational time can be reduced by using each occupancy rate per frequency band equal to each address of the lookup table as the characteristic of the input signal.
    • 一种使用多个查找表的数字信号编码方法和装置。 该方法包括:根据预定数量的地址中的输入信号的特性,准备多个查找表,该查找表存储用于编码输入信号频带的分配位数; 将时域中的输入信号划分成预定频带中的信号; 计算频带的地址值; 根据输入信号的特性选择多个查找表之一; 从所选择的查找表中提取相对于频带的具有所计算的地址值的地址的分配比特数,并将比特数分配给频带; 以及通过根据所分配的比特数来量化输入信号来生成比特流。 可以通过从根据输入信号的特性选择的最佳查找表中提取频带的分配比特数来执行适合于输入信号特性的比特量控制。 而且,通过使用等于查找表的每个地址的每个频带的每个占用率作为输入信号的特性,可以减少额外的计算时间。
    • 12. 发明申请
    • Lossless audio decoding/encoding method, medium, and apparatus
    • 无损音频解码/编码方法,介质和设备
    • US20050192799A1
    • 2005-09-01
    • US11066431
    • 2005-02-28
    • Junghoe KimMiao LeiShihwa LeeSangwook KimEnnmi OhDohyung Kim
    • Junghoe KimMiao LeiShihwa LeeSangwook KimEnnmi OhDohyung Kim
    • G10L19/02G10L19/00H04N5/60
    • G10L19/0017
    • A lossless audio encoding/decoding method, medium, and apparatus. The lossless audio encoding method includes converting an audio signal in a time domain into an audio spectral signal with an integer in a frequency domain, mapping the audio spectral signal in the frequency domain to a bit plane signal according to its frequency, and losslessly encoding binary samples of bit planes using a probability model determined according to a predetermined context. The lossless audio decoding method includes extracting a predetermined lossy bitstream and an error bitstream from error data by demultiplexing an audio bitstream, the error data corresponding to a difference between lossy encoded audio data and an audio spectral signal with an integer in a frequency domain, lossy decoding the extracted encoded lossy bitstream, losslessly decoding the extracted error bitstream, and restoring the original audio frequency spectral signal using the decoded lossy bitstream and error bitstream
    • 无损音频编码/解码方法,介质和装置。 无损音频编码方法包括将时域中的音频信号转换为频域中的整数的音频频谱信号,根据频率将频域中的音频频谱信号映射到比特平面信号,并将无损编码二进制 使用根据预定上下文确定的概率模型的位平面样本。 无损音频解码方法包括通过解复用音频比特流从误差数据中提取预定的有损比特流和错误比特流,对应于有损编码的音频数据与频域中的整数的音频频谱信号之间的差异的误差数据, 对所提取的编码有损比特流进行解码,对所提取的错误比特流进行无损解码,以及使用解码的有损比特流和误码比特流恢复原始音频频谱信号
    • 13. 发明授权
    • Scalable speech coding/decoding apparatus, method, and medium having mixed structure
    • 可扩展语音编码/解码装置,方法和具有混合结构的介质
    • US08271267B2
    • 2012-09-18
    • US11490139
    • 2006-07-21
    • Hosang SungSangwook KimRakesh TaoriKangeun Lee
    • Hosang SungSangwook KimRakesh TaoriKangeun Lee
    • G10L21/00
    • G10L19/24G10L19/0212G10L25/18
    • Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.
    • 提供了一种可扩展的宽带语音编码/解码装置,方法和媒体。 输入宽带语音输入信号首先被分成低频带信号和高频带信号。 然后使用码激励线性预测(CELP)方法对分频的低频带信号进行编码。 分频高频信号采用谐波法编码。 然后,使用修正的离散余弦变换(MDCT)方法对表示从低频带和高频带获得的合成信号之间的差异以及输入到低频带和高频带的信号进行编码的信号。 然后对编码信号进行多路复用。 然后输出复用的信号。 因此,可以实现对所有层的高质量语音。
    • 14. 发明授权
    • Method of and apparatus for encoding/decoding digital signal using linear quantization by sections
    • 使用线性量化部分对数字信号进行编码/解码的方法和装置
    • US07983346B2
    • 2011-07-19
    • US11125076
    • 2005-05-10
    • Junghoe KimDohyung KimShihwa LeeSangwook Kim
    • Junghoe KimDohyung KimShihwa LeeSangwook Kim
    • H04B14/04
    • G10L19/032
    • A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.
    • 提供了使用线性量化部分对数字信号进行编码/解码的方法及其装置。 编码方法包括:转换数字输入信号,从数字信号中去除冗余信息; 考虑数字信号的重要性,分配分配给每个预定量化单元的位数; 基于预定的量化单位将信号值的分布划分为预定的部分,并且线性量化数据转换为将数字输入信号逐段转换的操作; 以及从所述线性量化数据和预定侧信息生成比特流。 因此,与传统线性量化装置产生的声音质量相比,提高了音质,并且减少了非线性量化装置的复杂性。
    • 15. 发明申请
    • METHOD OF AND APPARATUS FOR ENCODING/DECODING DIGITAL SIGNAL USING LINEAR QUANTIZATION BY SECTIONS
    • 使用线性量化分段编码/解码数字信号的方法和装置
    • US20100239027A1
    • 2010-09-23
    • US12792048
    • 2010-06-02
    • Junghoe KIMDohyung KimShihwa LeeSangwook Kim
    • Junghoe KIMDohyung KimShihwa LeeSangwook Kim
    • H04B14/04
    • G10L19/032
    • A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.
    • 提供了使用线性量化部分对数字信号进行编码/解码的方法及其装置。 编码方法包括:转换数字输入信号,从数字信号中去除冗余信息; 考虑数字信号的重要性,分配分配给每个预定量化单元的位数; 基于预定的量化单位将信号值的分布划分为预定部分,以及在通过部分转换数字输入信号的操作中转换的线性量化数据; 以及从所述线性量化数据和预定侧信息生成比特流。 因此,与传统线性量化装置产生的声音质量相比,提高了音质,并且减少了非线性量化装置的复杂性。
    • 18. 发明申请
    • Low-bitrate encoding/decoding method and system
    • 低比特率编码/解码方法和系统
    • US20060004566A1
    • 2006-01-05
    • US11165569
    • 2005-06-24
    • Eunmi OhJunghoe KimSangwook KimAndrew EgorovAnton PorovKonstantin OsipovBoris Kudryashov
    • Eunmi OhJunghoe KimSangwook KimAndrew EgorovAnton PorovKonstantin OsipovBoris Kudryashov
    • G10L21/00
    • G10L19/0204G10L19/0017G10L19/032
    • A low-bitrate encoding system includes: a time-frequency transform unit transforming an input time-domain audio signal into a frequency-domain audio signal; a frequency component processor unit decimating frequency components in the frequency-domain audio signal; a psychoacoustic model unit modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; a quantizer unit quantizing the frequency-domain audio signal input from the frequency component processor unit to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit; and a lossless encoder unit encoding the quantized audio signal losslessly, and outputting the encoded audio signal in a bitstream format. Using the low-bitrate encoding system, it is possible to effectively compress data at a low bitrate, and thus to provide a high quality audio signal.
    • 低比特率编码系统包括:时间 - 频率变换单元,将输入的时域音频信号变换为频域音频信号; 频率分量处理器单元抽取频域音频信号中的频率分量; 心理声学模型单元,基于人类听觉特征对所接收的时域音频信号进行建模,以及计算编码比特分配信息; 量化器单元,根据从心理声学模型单元输入的编码比特分配信息量化从频率分量处理器单元输入的频域音频信号以具有比特率; 以及无损地编码量化音频信号的无损编码器单元,并以比特流格式输出编码音频信号。 使用低比特率编码系统,可以以低比特率有效地压缩数据,从而提供高质量的音频信号。
    • 19. 发明申请
    • Lossless audio coding/decoding method and apparatus
    • 无损音频编码/解码方法及装置
    • US20050203731A1
    • 2005-09-15
    • US11076284
    • 2005-03-10
    • Ennmi OhJunghoe KimMiao LeiShihwa LeeSangwook Kim
    • Ennmi OhJunghoe KimMiao LeiShihwa LeeSangwook Kim
    • H03M7/30G10L19/00G10L19/02H03M7/40
    • G10L19/0017
    • A lossless audio coding and/or decoding method and apparatus are provided. The coding method includes: mapping the audio signal in the frequency domain having an integer value into a bit-plane signal with respect to the frequency; obtaining a most significant bit and a Golomb parameter for each bit-plane; selecting a binary sample on a bit-plane to be coded in the order from the most significant bit to the least significant bit and from a lower frequency component to a higher frequency component; calculating the context of the selected binary sample by using significances of already coded bit-planes for each of a plurality of frequency lines existing in the vicinity of a frequency line to which the selected binary sample belongs; selecting a probability model by using the obtained Golomb parameter and the calculated contexts; and lossless-coding the binary sample by using the selected probability model. According to the method and apparatus, a compression ratio better than that of the bit-plane Golomb code (BPGC) is provided through context-based coding method having optimal performance.
    • 提供了一种无损音频编码和/或解码方法和装置。 编码方法包括:将具有整数值的频域中的音频信号映射到相对于频率的位平面信号中; 为每个位平面获取最高有效位和Golomb参数; 在从最高有效位到最低有效位以及从较低频率分量到较高频率分量的顺序中选择待编码的位平面上的二进制采样; 通过使用存在于所选择的二进制样本所属的频率线附近的多个频率线中的每一个的已经编码的位平面的重要性来计算所选二进制样本的上下文; 通过使用获得的Golomb参数和计算的上下文来选择概率模型; 并通过使用所选择的概率模型对二进制样本进行无损编码。 根据该方法和装置,通过具有最佳性能的基于上下文的编码方法提供比位平面Golomb码(BPGC)的压缩比更好的压缩比。