REDUCED COMPLEXITY CONVERTER SNR CALCULATION

    公开(公告)号:EP2917909B1

    公开(公告)日:2018-10-31

    申请号:EP13785889.0

    申请日:2013-11-04

    IPC分类号: G10L19/032

    摘要: The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for reducing the complexity of a bit allocation process used in the context of audio encoding/decoding. An audio encoder (300) configured to encode an audio signal according to a first audio codec system is described. The audio encoder (300) comprises a transform unit (302) configured to determine a set of spectral coefficients (312) based on the audio signal. Furthermore, the encoder (300) comprises a floating-point encoding unit (304) configured to determine a set of scale factors and a set of scaled values (314), based on the set of spectral coefficients (312); and to encode the set of scale factors to yield a set of encoded scale factors (313).

    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION
    5.
    发明公开
    COMPANDING APPARATUS AND METHOD TO REDUCE QUANTIZATION NOISE USING ADVANCED SPECTRAL EXTENSION 有权
    KOMPANDIERUNGSVORRICHTUNG和减少量化晚期频谱扩展程序

    公开(公告)号:EP2981963A1

    公开(公告)日:2016-02-10

    申请号:EP14720877.1

    申请日:2014-04-01

    摘要: 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 the 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.

    摘要翻译: 实施例涉及用于在音频编解码器中减少编码噪声压扩方法和系统。 的通过一个压缩processthat分割初始音频信号划分成使用所定义的windowshape段的多元化的初始音频信号的原始动态范围压缩处理减少,计算使用的基于非能量平均在频域中的宽带增益 初始音频信号的频域样本,并应用个别的增益值来放大相对低强度的段和衰减相对高强度的段。 然后,将压缩的音频信号扩展回原始动态范围内基本上没有适用逆增益值来放大相对高强度的段和相对低强度的衰减段。 甲QMF滤波器组来分析初始音频信号以获得频域表示。

    METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION
    6.
    发明公开
    METHOD AND SYSTEM FOR ENCODING AUDIO DATA WITH ADAPTIVE LOW FREQUENCY COMPENSATION 有权
    VERFAHREN UND SYSTEM ZUR KODIERUNG VON AUDIODATEN MIT ADAPTIVER NIEDRIGFREQUENZKOMPENSATION

    公开(公告)号:EP2803067A1

    公开(公告)日:2014-11-19

    申请号:EP12784365.4

    申请日:2012-09-25

    IPC分类号: G10L19/032 G10L19/02

    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尾数位分配的方法,包括通过对数据的一组低频带的每个频带执行自适应低频补偿。 所述低频补偿包括以下步骤:对所述音频数据执行音调检测,以产生指示所述组中的每个频带是否具有突出的音调内容的补偿控制数据; 并且对具有突出音调内容的集合中的每个频带执行低频补偿,包括通过校正具有突出音调内容的每个频带的初步屏蔽值,但是不对低频补偿中的任何其它频带中的音频数据进行低频补偿 组; 其中所述频域音频数据包括所述组的所述每个低频带的指数值,并且所述音调检测包括针对所述组的所述每个低频带确定所述音频的指数和对应的帐篷指数之间的差的度量 数据。 其他方面是包括这种音调检测和低频补偿步骤的音频编码方法,以及被配置为执行本发明方法的任何实施例的系统。

    SYSTEM FOR COMBINING LOUDNESS MEASUREMENTS IN A SINGLE PLAYBACK MODE
    7.
    发明公开
    SYSTEM FOR COMBINING LOUDNESS MEASUREMENTS IN A SINGLE PLAYBACK MODE 有权
    SYSTEM FOR组合体积测量在一个单一的WIEDERGABEMOUS

    公开(公告)号:EP2545646A1

    公开(公告)日:2013-01-16

    申请号:EP11707167.0

    申请日:2011-03-07

    IPC分类号: H03G9/00 H03G9/14

    CPC分类号: H03G9/00 H03G9/005 H03G9/14

    摘要: The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

    AUDIO ENCODING METHOD AND SYSTEM FOR GENERATING A UNIFIED BITSTREAM DECODABLE BY DECODERS IMPLEMENTING DIFFERENT DECODING PROTOCOLS

    公开(公告)号:EP2695162B1

    公开(公告)日:2018-08-22

    申请号:EP12714493.9

    申请日:2012-04-05

    IPC分类号: G10L19/16

    CPC分类号: G10L19/002 G10L19/167

    摘要: In a class of embodiments, an audio encoding system (typically, a perceptual encoding system that is configured to generate a single (“unified”) bitstream that is compatible with (i.e., decodable by) a first decoder configured to decode audio data encoded in accordance with a first encoding protocol (e.g., the multichannel Dolby Digital Plus, or DD+, protocol) and a second decoder configured to decode audio data encoded in accordance with a second encoding protocol (e.g., the stereo AAC, HE AAC v1, or HE AAC v2 protocol). The unified bitstream can include both encoded data (e.g., bursts of data) decodable by the first decoder (and ignored by the second decoder) and encoded data (e.g., other bursts of data) decodable by the second decoder (and ignored by the first decoder). In effect, the second encoding format is hidden within the unified bitstream when the bitstream is decoded by the first decoder, and the first encoding format is hidden within the unified bitstream when the bitstream is decoded by the second decoder. The format of the unified bitstream generated in accordance with the invention may eliminate the need for transcoding elements throughout an entire media chain and/or ecosystem. Other aspects of the invention are an encoding method performed by any embodiment of the inventive encoder, a decoding method performed by any embodiment of the inventive decoder, and a computer readable medium (e.g., disc) which stores code for implementing any embodiment of the inventive method.