摘要:
A parallel transfer rate converter inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1×(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2×(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.
摘要:
The subject disclosure is directed towards dynamically computing anti-aliasing filter coefficients for sample rate conversion in digital audio. In one aspect, for each input-to-output sampling rate ratio (pitch) obtained, anti-aliasing filter coefficients are interpolated based upon the pitch (e.g., using the fractional part of the ratio) from two filters (coefficient sets) selected based upon the pitch (e.g., using the integer part of the ratio). The interpolation provides for fine-grained cutoff frequencies, and by re-computation for each pitch, smooth anti-aliasing with dynamically changing ratios.
摘要:
A method for performing a spline interpolation to up-sample audio data includes up-sampling an input audio signal to generate a first signal having a first frequency. The input audio signal is sampled at an input frequency. The method also includes interpolating data of the first signal to generate a second signal having a second frequency. The data of the first signal is interpolated based on a B-spline interpolation function having at least a fourth order. The method includes down-sampling the second signal to generate an output audio signal having an output frequency. The method further includes updating a time index based on an integer operation that is immune to quantization error for a finite-word length implementation.
摘要:
It is known to perform sample rate conversion. A sample rate converter is arranged to receive digital data at an input sample rate Fs and to output data at an output sample rate Fo, where Fo=Fs/N, and N is decimation factor greater than 1. A problem can arise with sample rate converters when a user wishes to change the decimation rate. Generally a sample rate converter needs to discard the samples in its filter when the decimation rate is changed, and the filter output is unusable until the filter has refilled with values taken at the new decimation rate. The sample rate converter provided here does not suffer from this problem. The sample rate converter includes at least Q channels. Each channel comprises a Qth order filter arranged to select input signals at predetermined intervals from a run of P input signals, and to form a weighted sum of the selected input signals to generate an output value, and where the runs of P input signals of one channel are offset from the runs of P signals of the other channels.
摘要:
Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.
摘要:
The multi-branch rate change filter of the present invention achieves higher effective output rates by processing the input sample stream in two or more parallel filter branches with offset states.
摘要:
There is described a method of making a linear periodically time varying system shift-invariant, comprising the following steps implemented for each input signal the sampling rate of which has to be converted: —generating a set of polyphase components based on the input signal, —feeding the generated set of polyphase components to the system, and —generating an output signal by performing interleaving, shifting and addition on signals output by the system corresponding to the generated set of polyphase components processed by the system.
摘要:
The subject disclosure is directed towards a technology that may be used in an audio processing environment. Nodes of an audio flow graph are associated with virtual mix buffers. As the flow graph is processed, commands and virtual mix buffer data are provided to audio fixed-function processing blocks. Each virtual mix buffer is mapped to a physical mix buffer, and the associated command is executed with respect to the physical mix buffer. One physical mix buffer mix buffer may be used as an input data buffer for the audio fixed-function processing block, and another physical mix buffer as an output data buffer, for example.
摘要:
A system, method, and apparatus is disclosed for interpolation of an output of an analog to digital converter (ADC) to enable operation of the ADC at a sampling rate that is independent of the sampling rate for a DSP core so as to efficiently enable operation at higher date rates. According to one of the embodiments, an interpolation circuit is coupled between the ADC and DSP core and receives a first plurality of samples of data at the first data rate from the ADC and supplies a plurality of samples of second data at a second data rate to the DSP core; the second data rate being less than the first data rate. According to one of the embodiments, the interpolation circuit includes a memory and a FIR filter circuit having filter tap coefficient values selected to provide attenuation at high frequencies to reduce aliasing noise.
摘要:
Poly-phase filters are used to offer an efficient and low complexity solution to rate conversion. However, they suffer from inflexibility and are not easily reconfigured. A novel design for rate converters employ poly-phase filters but utilize interpolation between filter coefficients to add flexibility to rate conversion. This interpolation can be implemented as an interpolation of the poly-phase filter results. Additional approximations can be made to further reduce the amount of calculations required to implement a flexible rate converter.