METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION
    2.
    发明申请
    METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION 有权
    用自适应低频补偿编码音频数据的方法和系统

    公开(公告)号:US20140324441A1

    公开(公告)日:2014-10-30

    申请号:US14325130

    申请日:2014-07-07

    IPC分类号: G10L19/028 G10L19/26

    摘要: A method for determining mantissa bit allocation of audio data values of frequency domain audio data to be encoded. The allocation method includes a step of determining masking values for the audio data values, including by performing adaptive low frequency compensation on the audio data of each frequency band of a set of low frequency bands of the audio data. The adaptive low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set of low frequency bands has prominent tonal content; and performing low frequency compensation on the audio data in each frequency band in the set of low frequency bands having prominent tonal content as indicated by the compensation control data, but not performing low frequency compensation on the audio data in any other frequency band in the set of low frequency bands.

    摘要翻译: 一种用于确定要编码的频域音频数据的音频数据值的尾数位分配的方法。 分配方法包括通过对音频数据的一组低频带的每个频带的音频数据执行自适应低频补偿来确定音频数据值的屏蔽值的步骤。 所述自适应低频补偿包括以下步骤:对所述音频数据执行音调检测,以产生指示所述一组低频带中的每个频带是否具有突出的音调内容的补偿控制数据; 对由该补偿控制数据所表示的具有突出色调内容的低频带组中的每个频带中的音频数据执行低频补偿,而不对该组中的任何其它频带中的音频数据执行低频补偿 的低频带。

    ADAPTIVE BLOCK SWITCHING WITH DEEP NEURAL NETWORKS

    公开(公告)号:US20230386486A1

    公开(公告)日:2023-11-30

    申请号:US18248294

    申请日:2021-10-15

    摘要: The present invention relates to a method for predicting transform coefficients representing frequency content of an adaptive block length media signal, by receiving a frame and receiving block length information indicating a number of quantized transform coefficients for each block in the frame, the number of quantized transform coefficients being one of a first or second number, wherein the first number is greater than the second number, determining a first block has the second number of quantized transform coefficients, converting the first block into a converted block having the first number of quantized transform coefficients, conditioning a main neural network trained to predict at least one output variable given at least one conditioning variable, the at least one conditioning variable being based on information regarding the converted block and block length information for the first block, providing at least one predicted transform coefficients from an output stage of the main neural network.

    ADAPTIVE DIFFUSE SIGNAL GENERATION IN AN UPMIXER
    7.
    发明申请
    ADAPTIVE DIFFUSE SIGNAL GENERATION IN AN UPMIXER 有权
    自适应信号发生在自适应信号中

    公开(公告)号:US20160241982A1

    公开(公告)日:2016-08-18

    申请号:US15025074

    申请日:2014-09-26

    摘要: An audio processing system, such as an upmixer, may be capable of separating diffuse and non-diffuse portions of N input audio signals. The upmixer may be capable of detecting instances of transient audio signal conditions. During instances of transient audio signal conditions, the up-mixer may be capable of adding a signal-adaptive control to a diffuse signal expansion process in which M audio signals are output. The upmixer may vary the diffuse signal expansion process over time such that during instances of transient audio signal conditions the diffuse portions of audio signals may be distributed substantially only to output channels spatially close to the input channels. During instances of non-transient audio signal conditions, the diffuse portions of diffuse portions of audio signals may be distributed in a substantially uniform manner.

    摘要翻译: 诸如上混频器的音频处理系统可以能够分离N个输入音频信号的漫射和非漫射部分。 上混频器可能能够检测瞬态音频信号状况的实例。 在瞬态音频信号条件的情况下,上混频器可能能够对其中输出M个音频信号的扩散信号扩展处理添加信号自适应控制。 上升混频器可以随时间改变扩散信号扩展过程,使得在瞬态音频信号条件的情况下,音频信号的扩散部分可以基本上仅分布到空间上靠近输入信道的输出信道。 在非瞬态音频信号条件的情况下,音频信号的扩散部分的扩散部分可以以基本均匀的方式分布。

    Signal Decorrelation in an Audio Processing System
    10.
    发明申请
    Signal Decorrelation in an Audio Processing System 有权
    音频处理系统中的信号解相关

    公开(公告)号:US20150380000A1

    公开(公告)日:2015-12-31

    申请号:US14766371

    申请日:2014-01-22

    摘要: Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.

    摘要翻译: 音频处理方法可以涉及接收对应于多个音频频道的音频数据。 音频数据可以包括对应于音频编码或处理系统的滤波器组系数的频域表示。 解相关处理可以用音频编码或处理系统使用的相同的滤波器组系数执行。 可以在不将频域表示的系数转换到另一频域或时域表示的情况下执行去相关处理。 解相关过程可以涉及特定信道和/或特定频带的选择性或信号自适应去相关。 解相关过程可以包括将去相关滤波器应用于所接收的音频数据的一部分以产生滤波后的音频数据。 去相关处理可以包括使用非层级混合器来根据空间参数将接收的音频数据的直接部分与经滤波的音频数据组合。