Audio signal synthesis system based on probabilistic estimation of
time-varying spectra
    21.
    发明授权
    Audio signal synthesis system based on probabilistic estimation of time-varying spectra 有权
    基于时变谱概率估计的音频信号合成系统

    公开(公告)号:US06111183A

    公开(公告)日:2000-08-29

    申请号:US390918

    申请日:1999-09-07

    Applicant: Eric Lindemann

    Inventor: Eric Lindemann

    Abstract: The present invention describes methods and means for estimating the time-varying spectrum of an audio signal based on a conditional probability density function (PDF) of spectral coding vectors conditioned on pitch and loudness values. Using this PDF a time-varying output spectrum is generated as a function of time-varying pitch and loudness sequences arriving from an electronic music instrument controller. The time-varying output spectrum is converted to a synthesized output audio signal. The pitch and loudness sequences may also be derived from analysis of an input audio signal. Methods and means for synthesizing an output audio signal in response to an input audio signal are also described in which the time-varying spectrum of an input audio signal is estimated based on a conditional probability density function (PDF) of input spectral coding vectors conditioned on input pitch and loudness values. A residual time-varying input spectrum is generated based on the difference between the estimated input spectrum and the "true" input spectrum. The residual input spectrum is then incorporated into the synthesis of the output audio signal. A further embodiment is described in which the input and output spectral coding vectors are made up of indices in vector quantization spectrum codebooks.

    Abstract translation: 本发明描述了基于音调和响度值的频谱编码矢量的条件概率密度函数(PDF)来估计音频信号的时变频谱的方法和装置。 使用该PDF,产生随时间变化的音调和从电子乐器控制器到达的响度序列的函数的时变输出频谱。 时变输出频谱被转换为合成输出音频信号。 音调和响度序列也可以从输入音频信号的分析中导出。 还描述了用于响应于输入音频信号合成输出音频信号的方法和装置,其中基于条件概率密度函数(PDF),输入音频信号的时变频谱是基于条件概率密度函数 输入音高和响度值。 基于估计的输入光谱和“真实”输入光谱之间的差异产生残留的时变输入光谱。 然后将残留输入频谱合并到输出音频信号的合成中。 描述了另一实施例,其中输入和输出频谱编码矢量由矢量量化频谱码本中的索引组成。

    Continuous frequency dynamic range audio compressor
    22.
    发明授权
    Continuous frequency dynamic range audio compressor 失效
    连续频率动态范围音频压缩器

    公开(公告)号:US6097824A

    公开(公告)日:2000-08-01

    申请号:US870426

    申请日:1997-06-06

    CPC classification number: H04R25/453 H04R2430/03 H04R25/505

    Abstract: An improved multiband audio compressor is well behaved for both wide band and narrow band signals, and shows no undesirable artifacts at filter crossover frequencies. The compressor includes a heavily overlapped filter bank, which is the heart of the present invention. The filter bank filters the input signal into a number of heavily overlapping frequency bands. Sufficient overlapping of the frequency bands reduces the ripple in the frequency response, given a slowly swept sine wave input signal, to below about 2 dB, 1 dB, or even 0.5 dB or less with increasing amount of overlap in the bands. Each band is fed into a power estimator, which integrates the power of the band and generates a power signal. Each power signal is passed to a dynamic range compression gain calculation block, which calculates a gain based upon the power signal. Each band is multiplied by its respective gain in order to generate scaled bands. The scaled bands are then summed to generate an output signal.

    Abstract translation: 改进的多频带音频压缩器对于宽带和窄带信号都表现良好,并且在滤波器交叉频率处不显示不期望的伪像。 压缩机包括重叠的过滤器组,这是本发明的核心。 滤波器组将输入信号滤波成多个重叠的频带。 考虑到慢扫描的正弦波输入信号,频带的足够重叠使频带响应中的纹波减小到低于约2dB,1dB,甚至0.5dB或更小,随着频带中的重叠量增加。 每个频带被馈送到功率估计器中,该功率估计器对频带的功率进行积分并产生功率信号。 每个功率信号被传递到动态范围压缩增益计算块,其根据功率信号计算增益。 每个频带乘以其各自的增益,以便产生缩放的频带。 然后将缩放的频带相加以产生输出信号。

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