摘要:
Methods, digital systems, and computer readable media are provided for detection of music in an audio signal. Music is detected by partitioning the audio signal into overlapping frames, determining a fundamental frequency of a frame in the overlapping frames, including the fundamental frequency of the frame in a histogram of fundamental frequency values of frames occurring in the audio signal prior to the frame, and indicating that music is present in the audio signal when a number of occurrences of a fundamental frequency value in the histogram exceeds a threshold.
摘要:
Rebalancing of an audio signal refers to achieving a balance of perceived loudness, typically of right and left channels, given an unbalanced input. A flexible method to automatically rebalance an audio input signal is robust against noise in extreme cases through the individual channels combined in various ways as a function of the loudness ratio between input channels.
摘要:
Methods, digital systems, and computer readable media are provided for detection of music in an audio signal. Music is detected by partitioning the audio signal into overlapping frames, determining a fundamental frequency of a frame in the overlapping frames, including the fundamental frequency of the frame in a histogram of fundamental frequency values of frames occurring in the audio signal prior to the frame, and indicating that music is present in the audio signal when a number of occurrences of a fundamental frequency value in the histogram exceeds a threshold.
摘要:
Rebalancing of an audio signal refers to achieving a balance of perceived loudness, typically of right and left channels, given an unbalanced input. A flexible method to automatically rebalance an audio input signal is robust against noise in extreme cases through the individual channels combined in various ways as a function of the loudness ratio between input channels.
摘要:
Methods, digital systems, and computer readable media are provided for estimating change of amplitude and frequency in a digital audio signal by transforming a frame of the digital audio signal to the frequency domain, locating a frequency peak in the transformed frame, determining an interpolated peak of the located frequency peak, computing inner products of a portion of the transformed frame about the interpolated peak with a plurality of test signals, and estimating change of amplitude and change of frequency for the frequency peak from results of the inner products.
摘要:
Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.
摘要:
Methods, digital systems, and computer readable media are provided for determining a predominant fundamental frequency of a frame of an audio signal by finding a maximum absolute signal value in history data for the frame, determining a number of bits for downshifting based on the maximum absolute signal value, computing autocorrelations for the frame using signal values downshifted by the number of bits, and determining the predominant fundamental frequency using the computed autocorrelations.
摘要:
Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.
摘要:
Audio loudspeaker and headphone virtualizers and methods use room impulse responses with modified individual head-related transfer functions prior to superposition including middle truncation; and perform convolutions in the frequency domain with zero-padded sections to avoid circular convolution overlap.
摘要:
Methods, digital systems, and computer readable media are provided for determining a predominant fundamental frequency of a frame of an audio signal by finding a maximum absolute signal value in history data for the frame, determining a number of bits for downshifting based on the maximum absolute signal value, computing autocorrelations for the frame using signal values downshifted by the number of bits, and determining the predominant fundamental frequency using the computed autocorrelations.