Methods and Systems for Efficient Recovery of High Frequency Audio Content

    公开(公告)号:US20170221491A1

    公开(公告)日:2017-08-03

    申请号:US15494195

    申请日:2017-04-21

    Abstract: The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value for a first frequency subband of an audio signal is described. The first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value for the first frequency subband.

    Enhanced Block Switching and Bit Allocation for Improved Transform Audio Coding

    公开(公告)号:US20170178648A1

    公开(公告)日:2017-06-22

    申请号:US15378255

    申请日:2016-12-14

    Abstract: The present document relates to methods and apparatus for audio coding. In particular, the present document relates to methods and apparatus for enhanced block switching and/or bit allocation in audio coding of transient-tonal signals. A method of encoding samples of an audio signal comprises determining a first measure indicative of transient characteristics of the audio signal, determining a second measure indicative of tonal characteristics of the audio signal, selecting a transform length for the audio signal on the basis of the first measure and the second measure, and applying a time-frequency transform to a block of samples of the audio signal in accordance with the selected transform length, to thereby obtain a block of frequency coefficients corresponding to the block of samples of the audio signal. Another method of encoding samples of an audio signal comprises applying a time-frequency transform to the audio signal in accordance with a selected transform length, to thereby obtain a sequence of blocks of frequency coefficients, wherein each block of frequency coefficients among said sequence corresponds to a respective block of samples of the audio signal, determining a measure of tonal characteristics for a frequency band of the audio signal based on the blocks of frequency components among said sequence, selecting, for the blocks of frequency coefficients among said sequence, a quantization step size for the frequency coefficients in said frequency band on the basis of said measure of tonal characteristics, and quantizing, for the blocks of frequency coefficients among said sequence, the frequency coefficients in said frequency band in accordance with the selected quantization step size.

    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION
    6.
    发明申请
    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权
    使用高级光谱扩展降低量化噪声的装置和方法

    公开(公告)号:US20160019908A1

    公开(公告)日:2016-01-21

    申请号:US14762690

    申请日:2014-04-01

    Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

    Abstract translation: 实施例涉及用于减少音频编解码器中的编码噪声的压扩方法和系统。 压缩处理通过压缩处理来降低初始音频信号的原始动态范围,该压缩处理使用定义的窗口形状将初始音频信号分成多个段,使用基于非能量的平均值来计算频域中的宽带增益 初始音频信号的频域样本,并且应用各个增益值来放大相对较低强度的片段并衰减相对较高强度的片段。 压缩的音频信号然后被扩展回到基本上原始的动态范围,该动态范围应用反向增益值来放大相对较高强度的片段,并且衰减相对较低强度的段。 使用QMF滤波器组来分析初始音频信号以获得频域表示。

    PACKET LOSS CONCEALMENT
    8.
    发明公开

    公开(公告)号:US20230267938A1

    公开(公告)日:2023-08-24

    申请号:US18004197

    申请日:2021-07-07

    CPC classification number: G10L19/005 G10L19/008 G10L19/0204

    Abstract: Described are methods of processing an audio signal for packet loss concealment. The audio signal comprises a sequence of frames, each frame containing representations of a plurality of audio channels and reconstruction parameters for upmixing the plurality of audio channels to a predetermined channel format. One method includes: receiving the audio signal; and generating a reconstructed audio signal in the predefined channel format based on the received audio signal. Generating the reconstructed audio signal comprises: determining whether at least one frame of the audio signal has been lost; and if a number of consecutively lost frames exceeds a first threshold, fading the reconstructed audio signal to a predefined spatial configuration. Also described is a method of encoding an audio signal. Yet further described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

    METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION
    9.
    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

    CPC classification number: G10L19/028 G10L19/0204 G10L19/032 G10L19/265

    Abstract: 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.

    Abstract translation: 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。

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