A method for determining an inter-channel time difference (ITD) parameter includes determining a reference parameter according to a time-domain signal on a first sound channel and a time-domain signal on a second sound channel, where the reference parameter corresponds to a sequence of obtaining the time-domain signal on the first sound channel and the time-domain signal on the second sound channel, determining a search range according to the reference parameter and a limiting value (T.sub.max), where the T.sub.max is determined according to a sampling rate of the time-domain signal on the first sound channel, and performing search processing within the search range based on a frequency-domain signal on the first sound channel and a frequency-domain signal on the second sound channel to determine a first ITD parameter corresponding to the first sound channel and the second sound channel.

A method for processing a signal includes receiving a first frame of an input audio bitstream at a decoder. The first frame includes at least one signal associated with a frequency range. The method also includes decoding the at least one signal to generate at least one decoded signal having an intermediate sampling rate. The intermediate sampling rate is based on coding information associated with the first frame. The method further includes generating a resampled signal based at least in part on the at least one decoded signal. The resampled signal has an output sampling rate of the decoder.

A method for processing a signal includes receiving a first frame of an input audio bitstream at a decoder. The first frame includes at least one signal associated with a frequency range. The method also includes decoding the at least one signal to generate at least one decoded signal having an intermediate sampling rate. The intermediate sampling rate is based on coding information associated with the first frame. The method further includes generating a resampled signal based at least in part on the at least one decoded signal. The resampled signal has an output sampling rate of the decoder.

The invention relates to estimating tempo information directly from a bitstream encoding audio information, preferably music. Said tempo information is derived from at least one periodicity derived from a detection of at least two onsets included in the audio information. Such onsets are detected via a detection of long to short block transitions (in the bitstream) or/and via a detection of a changing bit allocation (change of cost) regarding encoding/transmitting the exponents of transform coefficients encoded in the bitstream.

Embodiments provide an audio processor for processing an audio signal to obtain a subband representation of the audio signal. The audio processor is configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal. Further, the audio processor is configured to perform a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.

Embodiments provide an audio processor for processing an audio signal to obtain a subband representation of the audio signal. The audio processor is configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal. Further, the audio processor is configured to perform a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.

The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.

The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length L.sub.a, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length L.sub.s, generating a frame of the output signal.

A method for processing a signal in the form of consecutive sample blocks, the method comprising filtering in a transformed domain of sub-bands, and particularly equalization processing, applied to a current block in the transformed domain, and filtering-adjustment processing that is applied in the transformed domain to at least one block adjacent to the current block.

The present technology relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program, configured to obtain a high quality audio with less encoding amount. A number-of-sections determining feature amount calculating circuit calculates a number-of-sections determining feature amount for determining the number of divisions to divide a process target section into continuous frame sections each including a frame for which the same estimation coefficient is selected, based on sub-band signals of a plurality of sub-bands constituting an input signal. A quasi-high frequency sub-band power difference calculating circuit determines the number of continuous frame sections in the process target section based on the number-of-sections determining feature amount, selects an estimation coefficient for obtaining a high frequency component of the input signal by estimation for each continuous frame section, and generates data including a coefficient index for obtaining the estimation coefficient. A high frequency encoding circuit encodes the obtained data, and generates high frequency encoded data. The present technology can be applied to an encoding device.