Commercially available active noise control headphones rely on fixed analog controllers to drive "anti-noise" loudspeakers. Our design uses an adaptive controller to optimally cancel unwanted acoustic noise. This headphone would be particularly useful for workers who operate or work near heavy machinery and engines because the noise is selectively eliminated. Desired sounds, such as speech and warning signals, are left to be heard clearly. The adaptive control algorithm is implemented on a Texas Instruments (TI™ )
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TMS320C30GEL digital signal processor (DSP), which drives a Sony CD550 headphone/microphone system. Our experiments indicate that adaptive noise control results in a dramatic improvement in performance over fixed noise control. This improvement is due to the availability of high-performance programmable DSPs and the self-optimizing and tracking
capabilities of the adaptive controller in response to the surrounding noise.
高的工作電壓高達100V N雙N溝道MOSFET同步驅動 The D810DCDC is a synchronous step-down switching regulator controller that can directly step-down voltages from up to 100V, making it ideal for telecom and automotive applications. The D810DCDC uses a constant on-time valley current control architecture to deliver very low duty cycles with accurate cycle-by-cycle current limit, without requiring a sense resistor. A precise internal reference provides 0.5% DC accuracy. A high bandwidth (25MHz) error amplifi er provides very fast line and load transient response. Large 1Ω gate drivers allow the D810DCDC to drive multiple MOSFETs for higher current applications. The operating frequency is selected by an external resistor and is compensated for variations in VIN and can also be synchronized to an external clock for switching-noise sensitive applications. Integrated bias control generates gate drive power from the input supply during start-up and when an output shortcircuit occurs, with the addition of a small external SOT23 MOSFET. When in regulation, power is derived from the output for higher effi ciency.
The Molgedey and Schuster decorrelation algorithm, having square mixing matrix and no noise . Truncation is used for the time shifted matrix, and it is forced to be symmetric . The delay Tau is estimated .
The number of independent components are calculated using Bayes Information Criterion (BIC), with PCA for dimension reduction.
Routine mar1psd: To compute the power spectum by AR-model parameters.
Input parameters:
ip : AR model order (integer)
ep : White noise variance of model input (real)
ts : Sample interval in seconds (real)
a : Complex array of AR parameters a(0) to a(ip)
Output parameters:
psdr : Real array of power spectral density values
psdi : Real work array
in chapter 12
Knowledge of the process noise covariance matrix
is essential for the application of Kalman filtering. However,
it is usually a difficult task to obtain an explicit expression of
for large time varying systems. This paper looks at an adaptive
Kalman filter method for dynamic harmonic state estimation and
harmonic injection tracking.
To estimate the input-output mapping with inputs x
％ and outputs y generated by the following nonlinear,
％ nonstationary state space model:
％ x(t+1) = 0.5x(t) + [25x(t)]/[(1+x(t))^(2)]
％ + 8cos(1.2t) + process noise
％ y(t) = x(t)^(2) / 20 + 6 squareWave(0.05(t-1)) + 3
％ + time varying measurement noise
％ using a multi-layer perceptron (MLP) and both the EKF and
％ the hybrid importance-samping resampling (SIR) algorithm.
Analytical constant-modulus algorithm, to separate linear combinations of CM sourcesThe algorithm
is robust in the presence of noise, and is tested on measured data,
collected from an experimental set-up.
Noncoherent receivers are attractive for pulsed UWB systems due to the implementation simplicity. To alleviate the noise effect in detecting UWB PPM signals, this letter proposes a simple yet flexible weighted noncoherent receiver structure, which adopts a square-law integrator multiplied with a window function.
