Audio encoding using adaptive codebook application ranges
    41.
    发明授权
    Audio encoding using adaptive codebook application ranges 有权
    使用自适应码本应用范围的音频编码

    公开(公告)号:US09361894B2

    公开(公告)日:2016-06-07

    申请号:US13895256

    申请日:2013-05-15

    Inventor: Yuli You

    Abstract: A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal.

    Abstract translation: 低比特率数字音频编码系统包括:编码器,其基于其本地属性将码本分配给量化索引组,导致独立于块量化边界的码本应用范围。 本发明还包括分辨率滤波器组或三模式分辨率滤波器组,其可以在高频和低频分辨率模式或高,低和中等模式之间选择性地切换,例如当检测到帧中的瞬态时。 结果是具有用于有效传输或存储的显着较低比特率的多声道音频信号。 解码器本质上是编码器的结构和方法的逆,并且导致不能与原始信号可听区分的再现音频信号。

    Method and Arrangement for Scalable Low-Complexity Coding/Decoding
    44.
    发明申请
    Method and Arrangement for Scalable Low-Complexity Coding/Decoding 有权
    可扩展低复杂度编码/解码的方法和布置

    公开(公告)号:US20150149161A1

    公开(公告)日:2015-05-28

    申请号:US14405707

    申请日:2012-11-13

    CPC classification number: G10L19/035 G10L19/002 G10L19/08

    Abstract: In a quantization method for quantizing a received excitation signal in a communication system performing the steps of re-shuffling S301 the elements of the received excitation signal to provide a re-shuffled excitation signal; coding S302 the re-shuffled excitation signal with a variable bit-rate algorithm to provide a coded excitation signal; and reassigning S303 codewords of the coded excitation signal if a number of used bits exceeds a predetermined fixed bit rate requirement to provide a quantized excitation signal.

    Abstract translation: 在用于量化接收到的激励信号的量化方法中,执行以下步骤的通信系统中的接收到的激励信号:重新混洗S301所接收的激励信号的元素以提供重新混洗的激励信号; 用可变比特率算法编码S302重新混洗的激励信号以提供编码的激励信号; 并且如果所使用的比特数超过预定的固定比特率要求以提供量化的激励信号,则重新分配编码的激励信号的S303码字。

    Audio encoder and decoder
    45.
    发明授权
    Audio encoder and decoder 有权
    音频编码器和解码器

    公开(公告)号:US08924201B2

    公开(公告)日:2014-12-30

    申请号:US13901960

    申请日:2013-05-24

    CPC classification number: G10L19/26 G10L19/008 G10L19/032 G10L19/035

    Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

    Abstract translation: 本发明教导了一种新的音频编码系统,其可以以低比特率良好地对一般音频和语音信号进行编码。 所提出的音频编码系统包括用于基于自适应滤波器对输入信号进行滤波的线性预测单元; 变换单元,用于将经滤波的输入信号的帧变换为变换域; 以及用于量化变换域信号的量化单元。 量化单元基于输入信号特性来决定用基于模型的量化器或非基于模型的量化器对变换域信号进行编码。 优选地,该决定基于由变换单元应用的帧大小。

    Signal encoding apparatus, signal encoding method, and program
    46.
    发明申请
    Signal encoding apparatus, signal encoding method, and program 有权
    信号编码装置,信号编码方法和程序

    公开(公告)号:US20050075872A1

    公开(公告)日:2005-04-07

    申请号:US10500103

    申请日:2002-12-25

    Abstract: The present invention intends to render quantization noise virtually imperceptible for a user and to prevent reduction in frequency resolution and reduction in encoding efficiency. A signal encoding apparatus includes: a quantization unit for quantizing an input signal based on a plurality of quantization methods; a dequantization unit for obtaining decoded signals by performing the dequantizing process; an error signal calculation unit for calculating a plurality of error signals between the decoded signals and the input signal; a weighting calculation unit for calculating, for each subblock, a weight related to degree concerning whether or not quantization noise corresponding to error signal is virtually imperceptible for a user; a quantization method selection unit for selecting a given quantization method from among the plurality of quantization methods, when a plurality of weighted error signals, obtained by assigning a weight of each subblock to an error signal of the subblock, are generated, based on the of weighted error signals; and an output unit for outputting the input signal quantized based on the given quantization method as an output signal.

    Abstract translation: 本发明意图使得量化噪声对于用户实际上是不可察觉的,并且防止频率分辨率的降低和编码效率的降低。 信号编码装置包括:量化单元,用于基于多个量化方法量化输入信号; 去量化单元,用于通过执行去量化处理来获得解码信号; 误差信号计算单元,用于计算解码信号与输入信号之间的多个误差信号; 一个加权计算单元,用于针对每个子块计算与用户对于误差信号对应的量化噪声是否几乎不可察觉的程度相关的权重; 一种用于从多个量化方法中选择给定的量化方法的量化方法选择单元,当通过将每个子块的权重分配给子块的误差信号而获得的多个加权误差信号基于 加权误差信号; 以及输出单元,用于输出基于给定量化方法量化的输入信号作为输出信号。

    High efficiency audio encoding method and apparatus
    47.
    发明授权
    High efficiency audio encoding method and apparatus 失效
    高效音频编码方法及装置

    公开(公告)号:US5832426A

    公开(公告)日:1998-11-03

    申请号:US568481

    申请日:1995-12-07

    CPC classification number: H04H20/88 H04B1/665

    Abstract: A method and apparatus for high efficiency encoding of digital data whereby input digital data may be encoded by so-called high efficiency encoding. The high efficiency device includes a signal component encoding circuit including in turn a bandwidth storage circuit 62 for holding the information on the playback band of a previous block, that is the information on the bandwidth of bit allocation performed in the previous block, or the value of the number of the playback encoding units, and a control circuit 63 for deciding the number of playback encoding units of the current block based upon the playback band information, that is the value of the number of the playback encoding units, in the previous block, held by the bandwidth storage circuit 62. Since the stable playback bandwidth is maintained, and there is no risk of the playback band being frequently changed from one block to another, the harsh sounding noise otherwise produced by the appearance and disappearance of high-range side signals is not produced, so that deterioration in the perceived sound quality may be suppressed to a minimum with an insufficient bit rate.

    Abstract translation: 一种用于数字数据的高效编码的方法和装置,由此可以通过所谓的高效率编码对输入数字数据进行编码。 高效率器件包括一个信号分量编码电路,该信号分量编码电路又包括一个带宽存储电路62,用于保存关于先前块的重放带上的信息,即前一个块中执行的比特分配带宽的信息, 以及控制电路63,用于根据重放频带信息来确定当前块的重放编码单元的数量,即重放编码单元的数量在前一块中 由带宽存储电路62保持。由于保持了稳定的重放带宽,并且没有再现频带从一个块频繁地变化到另一个块的风险,否则由高范围的外观和消失产生的恶劣的声音噪声 不会产生侧面信号,从而可以以不足的比特率将感知到的声音质量的劣化抑制到最小。

    Apparatus and method for harmonicity-dependent tilt control of scale parameters in an audio encoder

    公开(公告)号:US20240371382A1

    公开(公告)日:2024-11-07

    申请号:US18307535

    申请日:2023-04-26

    Abstract: A method and an apparatus for encoding an audio signal. The apparatus includes a converter converting the audio signal into a spectral representation; a scale parameter calculator calculating scale parameters; a spectral processor processing the spectral representation using the scale parameters; and a scale parameter encoder generating an encoded representation of the scale parameters. The scale parameter calculator is calculates an amplitude-related measure for each band to obtain a set of amplitude-related measures. A pre-emphasis operation is performed to the amplitude-related measures, so that low frequency amplitudes are emphasized with respect to high frequency amplitudes according to a tilt value, or a pre-emphasis factor. The scale parameter calculator controls the tilt value, or the pre-emphasis factor, based on a harmonicity measure of the audio signal.

    Optimised spherical vector quantisation
    49.
    发明公开

    公开(公告)号:US20240304198A1

    公开(公告)日:2024-09-12

    申请号:US18570904

    申请日:2022-07-05

    Applicant: ORANGE

    CPC classification number: G10L19/035 G10L19/008 G10L19/038

    Abstract: A method for encoding an input point on an n-dimensional sphere by encoding n-1 spherical coordinates of said input point. The method includes sequential scalar quantization of the n-1 spherical coordinates in order to obtain at most 2n-2 candidates at the end of the sequential scalar quantization of the n-1 coordinates, and subsequently selecting the best candidate which minimizes a distance between the input point and the at most 2n-2 candidates, and determining the separate quantization indices resulting from the sequential scalar quantization of the spherical coordinates of the best candidate and sequentially encoding the separate quantization indices of the best candidate. A corresponding decoding method, an encoding device and a decoding device are also provided.

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