The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank (501) comprising an analysis transformation unit (601) having a frequency resolution of Δf; and an analysis window (611) having a duration of D.sub.A; the analysis filter bank (501) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit (502, 650) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank (504) comprising a synthesis transformation unit (602) having a frequency resolution of QΔf; and a synthesis window (612) having a duration of D.sub.s; the synthesis filter bank (504) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≥1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration D.sub.A of the analysis filter bank is selected based on the frequency resolution factor Q.

Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element. The next elements(s) can, for example, be stored in an input buffer, then decoded and stored in a decoded sample buffer, all prior to the required presentation time of the multiple element effect. At the requisite time, a client device component can access the respective samples of the decoded audio clips as it performs the cross-fade, mix or other effect. Such exemplary embodiments use a single decoder and thus do not require synchronized simultaneous decodes.

Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element. The next elements(s) can, for example, be stored in an input buffer, then decoded and stored in a decoded sample buffer, all prior to the required presentation time of the multiple element effect. At the requisite time, a client device component can access the respective samples of the decoded audio clips as it performs the cross-fade, mix or other effect. Such exemplary embodiments use a single decoder and thus do not require synchronized simultaneous decodes.

A system for using an audio watermark to avoid double processing. The decoder inserts the audio watermark during a transient in the audio signal. This avoids the drawbacks of using out-of-band control signals or metadata. The decoder performs detecting a transient in a first audio signal, transforming a portion related to the transient into the frequency domain to compare a first band of the frequency domain data and a second band of the frequency domain data, when the first band is uncorrelated with the second band, the decoder performs processing on the first audio data to generate a second audio data, when the first band is correlated, the first audio data is used as the second audio data without performing any processing.

Performance of an encoding process and a decoding process for a sound signal is enhanced. A representative value calculating part 110 calculates, for each frequency section by a plurality of samples fewer than the number of frequency samples of a sample sequence of a frequency domain signal corresponding to an input acoustic signal, from the sample sequence of the frequency domain signal, a representative value of the frequency section from sample values of samples included in the frequency section, for each of predetermined time sections. A signal companding part 120 obtains, for each of the predetermined time sections, a frequency domain sample sequence obtained by multiplying a weight according to a function value of the representative value by a companding function for which an inverse function can be defined and each of the samples corresponding to the representative value in the sample sequence of the frequency domain signal, as a sample sequence of a weighted frequency domain signal.

Provided is pitch enhancement processing having little unnaturalness even in time segments for consonants, and having little unnaturalness to listeners caused by discontinuities even when time segments for consonants and other time segments switch frequently. A pitch emphasis apparatus carries out the following as the pitch enhancement processing: for a time segment in which a spectral envelope of a signal has been determined to be flat, obtaining an output signal for each of times in the time segment, the output signal being a signal including a signal obtained by adding (1) a signal obtained by multiplying the signal of a time, further in the past than the time by a number of samples T.sub.0 corresponding to a pitch period of the time segment, a pitch gain σ.sub.0 of the time segment, a predetermined constant B.sub.0, and a value greater than 0 and less than 1, to (2) the signal of the time.

An audio encoder for providing an output signal using an input audio signal includes a patch generator, a comparator and an output interface. The patch generator generates at least one bandwidth extension high-frequency signal, wherein a bandwidth extension high-frequency signal includes a high-frequency band. The high-frequency band of the bandwidth extension high-frequency signal is based on a low frequency band of the input audio signal. A comparator calculates a plurality of comparison parameters. A comparison parameter is calculated based on a comparison of the input audio signal and a generated bandwidth extension high-frequency signal. Each comparison parameter of the plurality of comparison parameters is calculated based on a different offset frequency between the input audio signal and a generated bandwidth extension high-frequency signal. Further, the comparator determines a comparison parameter from the plurality of comparison parameters, wherein the determined comparison parameter fulfils a predefined criterion.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

An audio signal encoding method and apparatus, and an audio signal decoding method and apparatus are provided. The audio signal encoding method includes: obtaining a frequency-domain coefficient of a current frame and a frequency-domain coefficient of a reference signal of the current frame; performing filtering processing on the frequency-domain coefficient of the current frame to obtain a filtering parameter; determining a target frequency-domain coefficient of the current frame based on the filtering parameter; performing filtering processing on the frequency-domain coefficient of the reference signal and a reference frequency-domain coefficient based on the filtering parameter to obtain a target frequency-domain coefficient of the reference signal; and encoding the target frequency-domain coefficient of the current frame based on the target frequency-domain coefficient of the current frame, the target frequency-domain coefficient of the reference signal, a reference target frequency-domain coefficient. The method can improve audio signal encoding/decoding efficiency.