A method, a computer-readable medium, and an apparatus for resampling audio signal are provided. The apparatus resamples the audio signal in order to preserve the audio playback quality when dealing with audio playback overrun and underrun problem. The apparatus may receive a data block of the audio signal including a first number of samples. For each sample of the first number of samples, the apparatus may slice a portion of the audio signal corresponding to the sample into a particular number of sub-samples. The apparatus may resample the data block of the audio signal into a second number of samples based on the first number of samples and the particular number of sub-samples associated with each sample of the first number of samples. The apparatus may play back the resampled data block of the audio signal via an electroacoustic device.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

Disclosed herein is a method and apparatus for converting a stream of samples at a first sampling rate to a stream of samples at a second sampling rate. An exemplary method includes measuring the first sampling rate; determining a first upsampling factor from a basis including: the measured first sampling rate, the target value of the second sampling rate, and a resynchronisation error factor, the first upsampling factor being constrained to be an integer power of a predetermined integer value; and deriving, from a reference set of filter coefficients and from a ratio of the first upsampling factor to a reference upsampling factor, a first set of filter coefficients for use in a first interpolation filter, the reference set of filter coefficients being for a reference upsampling factor that is an integer power of the predetermined integer value.

The present disclosure provides a resampling apparatus and a resampling method. The resampling apparatus includes a control unit, a memory device, a resolution identifier, a phase rate generator, a coefficient generator, and a resample filter. The control unit controls reading and writing operations of the resampling apparatus according to a control signal. The memory device transmits the control signal to the control unit. The resolution identifier sets a resolution bandwidth identity according to an interpolation/decimation (I/D) value of the control signal. The phase rate generator generates a phase select signal and a counter enable signal according to the resolution bandwidth identity. The coefficient generator generates a coefficient select signal according to the resolution bandwidth identity. The resample filter generates a resampled output data according to the phase select signal, the coefficient select signal, and an input data.

A method of converting a stream of samples at a first sampling rate to a stream of samples at a second sampling rate is disclosed, comprising: measuring the first sampling rate; determining a first upsampling factor from a basis comprising: the measured first sampling rate, the target value of the second sampling rate, and a resynchronisation error factor, the first upsampling factor being constrained to be an integer power of a predetermined integer value; and deriving, from a reference set of filter coefficients and from a ratio of the first upsampling factor to a reference upsampling factor, a first set of filter coefficients for use in a first interpolation filter, the reference set of filter coefficients being for a reference upsampling factor that is an integer power of the predetermined integer value.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

An oversampling channelizer for processing overlapping data that includes a data storage unit, coupled to a data line that receives data values. The data storage unit includes a plurality of lanes, wherein each of the plurality of lanes includes dedicated memory locations and wires that store and transmit data values for a data vector of a data frame, and that store and transmit additional data values for a subsequent data vector of a subsequent data frame that includes a plurality of the data values from the data vector in the data frame. The oversampling channelizer includes a coefficient storage unit that stores a plurality of coefficient vectors for a plurality of coefficient frames. The oversampling channelizer includes a computation unit that computes a dot product of the data values for the data vectors of the data frame with coefficient values for coefficient vectors of a coefficient frame selected by a coefficient storage unit.