A signal conversion from an input signal to an output signal where the filter used is factorized so that the conversion comprises determining 1) only a first factor at each sampling time of the input signal, where this first factor is independent on the sampling times of the output signal, and 2) only a second factor at each sampling time of the output signal, where this second factor is independent of the sampling times of the input signal. This reduces the computational load for this conversion. In addition, for most filters, the factors may be calculated recursively further increasing the computational load and also reducing the storage requirements. This allows for instantaneous changes in the sampling rates or non-uniform sampling rates with low computational requirements and low memory usage.

Disclosed herein, among other things, are methods and apparatus for providing a time-stamp based controller for synchronization of sink or source sampling rate with external packet rate. A method for wireless communications includes receiving a transmission of a packet using a wireless transceiver of an electronic device, and using a processor of the electronic device to read a first value of a system timer and store the first value as an arrival time-stamp. The packet is decoded and processed by the processor, and sent to an output. When the processed packet is sent, a second value of the system timer is read, adjusted and stored as a departure time-stamp. The arrival time-stamp and the departure time-stamp are used to calculate an adjustment stimulus for a sample rate actuator of the electronic device. The sample rate actuator is configured to maintain synchronization of sampling rate with an external packet rate.

A resampling method based on window function for flexible sampling rate conversion in broadband frequency measurement devices is described. The resampling algorithm can satisfy the requirements of different sampling rates. The frequency responses of the filter in the resampling model based on the Farrow structure are analyzed, and the design criterion of the filter in resampling model is considered. A fractional delay filter design model based on window function method is described. A fractional delay filter matrix, which is expressed by polynomial form, is constructed. Then the expression related to subfilter coefficients is obtained and subfilter coefficients are solved for by the least square method.

An instrument configured to process signal data is disclosed. The instrument is operable to control and or change the sampling rate of the signal data from a first sample rate to a second sample rate different than the first sample rate.

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.

A resampling method based on window function for flexible sampling rate conversion in broadband frequency measurement devices is described. The resampling algorithm can satisfy the requirements of different sampling rates. The frequency responses of the filter in the resampling model based on the Farrow structure are analyzed, and the design criterion of the filter in resampling model is considered. A fractional delay filter design model based on window function method is described. A fractional delay filter matrix, which is expressed by polynomial form, is constructed. Then the expression related to subfilter coefficients is obtained and subfilter coefficients are solved for by the least square method.

Methods and systems for finite impulse response filter under a constrained sampling rate. A method for constrained sampling rate filtering includes segmenting an input signal with a first rate into a defined number of streams, wherein the defined number of streams sets an effective sample rate of each stream to at least less than a second rate and the second rate is less than the first rate, determining filter coefficients for a finite impulse response filter, grouping the filter coefficients into subsets of filter coefficients to match the defined number of streams, applying in parallel, for each of the defined number of streams, a subset of filter coefficients to a corresponding stream, and combining at least some outputs from the defined number of streams conditionally based on the second rate.

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.

An instrument configured to process signal data is disclosed. The instrument is operable to control and or change the sampling rate of the signal data from a first sample rate to a second sample rate different than the first sample rate.

An instrument configured to process signal data is disclosed. The instrument is operable to control and or change the sampling rate of the signal data from a first sample rate to a second sample rate different than the first sample rate.