摘要:
Systems and methods for converting a digital input data stream from a first sample rate to a second, fixed sample rate using a combination of hardware and software components. In one embodiment, a system includes a rate estimator configured to estimate the sample rate of an input data stream, a phase selection unit configured to select a phase for interpolation of a set of polyphase filter coefficients based on the estimated sample rate, a coefficient interpolator configured to interpolate the filter coefficients based on the selected phase, and a convolution unit configured to convolve the interpolated filter coefficients with samples of the input data stream to produce samples of a re-sampled output data stream. One or more hardware or software components are shared between multiple channels that can process data streams having independently variable sample rates.
摘要:
Systems and methods for converting a digital input data stream from a first sample rate to a second, fixed sample rate using a combination of hardware and software components. In one embodiment, a system includes a rate estimator configured to estimate the sample rate of an input data stream, a phase selection unit configured to select a phase for interpolation of a set of polyphase filter coefficients based on the estimated sample rate, a coefficient interpolator configured to interpolate the filter coefficients based on the selected phase, and a convolution unit configured to convolve the interpolated filter coefficients with samples of the input data stream to produce samples of a re-sampled output data stream. One or more hardware or software components are shared between multiple channels that can process data streams having independently variable sample rates.
摘要:
Systems and methods for converting a data stream from a first sample rate to a second sample rate, where the data is received in bursts. In one embodiment, a method includes receiving bursty audio data on a first input line and receiving synchronization data on a second input line that is separate from the first input line. An input sample rate is then estimated for the received audio data based on the received synchronization data and the audio data is converted to an output sample rate. The input sample rate is determined by counting samples received in a time interval and potentially low-pass filtering the result. The audio data may be in packetized, parallel, or other forms, and the synchronization data may include individual signals, such as pulses or bits received at regular or irregular intervals.
摘要:
Systems and methods are provided for converting input data streams having variable input sample rates to an output sample rate, which systems and methods are used in processing the data streams. In one embodiment, a system includes a clock source, a counter configured to count cycles for a corresponding data stream, and a data processor. The data processor is configured to read the number of cycles counted by the counter between received frame sync signals and to convert the first data stream to the predetermined output sample rate based on the corresponding number of cycles counted.
摘要:
Systems and methods for performance improvements in digital switching amplifiers using a low delay corrector. In one embodiment, a digital pulse width modulation (PWM) amplifier includes a signal processing plant configured to receive and process an input audio signal. The amplifier also includes a low delay corrector configured to receive signals output by the plant. The output of the low delay corrector is added to the input audio signal as feedback. The plant may consist of a modulator and power switch, a noise shaper, or any other type of plant. An analog-to-digital converter (ADC) may be provided to convert the output audio signal to a digital signal. Filtering may be implemented before or after the ADC, and a decimator may be placed after the ADC if it is an oversampling ADC.
摘要:
Systems and methods for ensuring proper phase alignment of audio signals which are processed by separate hardware channels in an audio amplification system. In one embodiment, the phase alignment is controlled by determining the number of audio data samples which are stored in the input buffers of multiple audio amplification units and controlling reads from the input buffers to minimize the difference between an actual read-write pointer differential and a target differential. In a master unit, the target differential is a predetermined target value corresponding to a desired delay in the buffer. The actual pointer differential of the master unit is passed to one or more slave units. The actual pointer differential of the master unit is used as the target differential of the slave units. The pointer differentials of the slave units are thereby driven to track the pointer differential of the master unit, keeping the units synchronized.
摘要:
Systems and methods for using multiple rate estimate counters in converting input data streams having variable sample rates to an output sample rate that are used in processing the data streams. In one embodiment, a system includes a clock source, first and second counters coupled to the clock source and configured to count cycles for corresponding data streams, and a data processor coupled to the first and second counters. The data processor is configured to read the number of cycles counted by each of the counters between received frame sync signals and to convert the first data stream to the predetermined output sample rate based on the corresponding number of cycles counted, and to convert the second data stream to the predetermined output sample rate based on the ratio of the numbers of cycles counted in the first and second counters.
摘要:
Systems and methods for performance improvements in digital switching amplifiers using simulation-based feedback. In one embodiment, a digital pulse width modulation (PWM) amplifier includes a signal processing plant configured to receive and process an input audio signal. The amplifier also includes a simulator configured to model processing of audio signals by the plant. The outputs of the plant and the simulator are provided to a subtractor, the output of which is then added to the input audio signal as feedback. The plant may consist of a modulator and power switch, a noise shaper, or any other type of plant. An analog-to-digital converter (ADC) may be provided to convert the output audio signal to a digital signal for input to the subtractor. Filtering may be implemented before or after the ADC, and a decimator may be placed after the ADC if it is an oversampling ADC.
摘要:
A low delay corrector (LDC) unit includes a non-linear function generator and a filter. The nonlinear function generator receives a first signal and outputs a second signal in dependence on the first signal and a transfer function of the nonlinear function generator. The filter is fed in dependence on the second signal output by the nonlinear function generator. The first signal received by the nonlinear function generator is derived in dependence on an input signal provided to an input of the LDC unit and an output of the filter. An output of the LDC unit is derived in dependence on the first signal received by the nonlinear function generator and the second signal output by the nonlinear function generator.
摘要:
Systems and methods for performance improvements in digital switching amplifiers using low-pass filtering to reduce noise and distortion. In one embodiment, a digital pulse width modulation (PWM) amplifier includes a signal processing plant configured to receive and process an input audio signal. The amplifier also includes a low-pass filter configured to filter audio signals output by the plant. The filtered output of the plant is added to the input audio signal as feedback. The plant may consist of a modulator and power switch, a noise shaper, or any other type of plant. An analog-to-digital converter (ADC) may be provided to convert the output audio signal to a digital signal. Filtering may be implemented before or after the ADC, and a decimator may be placed after the ADC if it is an oversampling ADC.