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MT Downconvert Passband (Real) (G Dataflow)

Last Modified: January 9, 2017

Downconverts real passband signal data of a user-specified bandwidth. This node can be used in simulated as well as hardware-equipped applications.



Signal for downconversion in passband form.


carrier frequency

The center frequency of the passband, in hertz (Hz). This frequency is downconverted to 0 Hz. Enter the expected carrier frequency of the incoming signal for downconversion.


passband bandwidth

The bandwidth, in Hz, of the passband signal data. The node ignores this parameter if you set the reset? parameter to FALSE.


advanced filter parameters

Filter parameters.


passband ripple

The ripple in the passband, in dB. The ripple is the ratio of the maximum deviation from the average passband amplitude to the average passband amplitude. The value must be greater than zero.

Default: 0.01


stopband start

The start of the stopband, Hz.

Default: 0


compute filter stopband ?

A Boolean that determines whether to compute the stopband start or use the value that you specify in the stopband start parameter.

TRUE Computes the stopband start based on the carrier frequency and passband bandwidth parameters.
FALSE Uses the value that you specify in the stopband start parameter.

Default: TRUE


stopband gain

The stopband gain, in dB. The gain is the negative of the minimum attenuation of the stopband with respect to the average amplitude of the passband.

Default: -96


error in

Error conditions that occur before this node runs. The node responds to this input according to standard error behavior.

Default: no error


enable filter

A Boolean that determines whether to perform software filtration on the downconverted data.

TRUE The node filters the downconverted waveform parameter using a software FIR filter.
FALSE Disables the downconversion filter and generates unfiltered data in the downconverted waveform parameter.

Default: TRUE



A Boolean that determines whether to use values specified by the initial phase, passband bandwidth, enable filter, passband ripple, and stopband start parameters.

TRUE The node uses these parameter values at each call.
FALSE The node ignores these parameters and continues using values supplied in the previous call. Reusing previous input values is useful when sequential data blocks represent contiguous signal data.

Default: FALSE


initial phase

The initial phase, in degrees, of the software local oscillator used in the downconversion process. The node ignores this parameter if you set the reset? parameter to FALSE. Use the initial phase parameter to match the phase of the incoming modulated carrier and the local oscillator(s) of the downconversion process.


downconverted waveform

The downconverted signal in complex envelope format.



The trigger (start) time of the acquired signal.



Time interval between data points in the acquired signal.



The complex-valued time-domain data array. The real and imaginary parts of this complex data array correspond to the in-phase (I) and quadrature-phase (Q) data, respectively.



The deviation of the passband gain from the nominal gain of 0 dB.


filter length

Number of taps in the filter.


error out

Error information. The node produces this output according to standard error behavior.

Downconversion Filter

Using a downconversion filter with this node may lead to the apparent loss of samples in the output waveform. Refer to Filter Delay for more information about this effect. When you set enable filter to TRUE, the downconverted I/Q signal is filtered by a software lowpass FIR filter with design parameters that are dictated by the following considerations:
  • If carrier frequency is greater than passband bandwidth, the filter stopband begins at carrier frequency.
  • If carrier frequency is less than passband bandwidth, the filter stopband begins between carrier frequency and (2 * carrier frequency) - (passband bandwidth/2).

Filter Delay

Finite impulse response (FIR) filters are used for different operations such as pulse-shaping, matched filtering, and downconversion filtering. For such filters, the output signal is related to the input signal as shown by the following equation:
y [ n ] = b 0 x [ n ] + b 1 x [ n 1 ] + ... + b P x [ n P ]


P is the filter order

x[n] is the input signal

y[n] is the output signal

bi are the filter coefficients

The initial state for all samples in an FIR filter is 0. The filter output until the first input sample reaches the middle tap (the first causal sample) is called the transient response, or filter delay. For an FIR filter that has N taps, the delay is (N-1)/2 samples. This relationship is illustrated in the following figure, where a sine wave is filtered by an FIR filter with 50 taps.

Recovering Samples in Single-Shot Operations

In single-shot operations for modulators and demodulators, the filter delay is truncated before the signal is generated because these samples are not valid. Some samples at the end of the block do not appear at the modulator or demodulator output, and hence appear to have been lost.

You can recover these samples by sending extra samples to the modulator or demodulator. To determine how many extra samples you must add, use the following guidelines:
  • For modulation: Let L be the pulse-shaping filter length, m be the number of samples per symbol, and M be the modulation order. The number of bits to be added to the input bit stream is given by the following formula:
    N = ( L 1 ) log 2 M m
  • For demodulation: Demodulation use filters during matched filtering. Let L be the length of the matched filter. The number of samples to be added to the input signal prior to filtering is given by the following formula:
    N = L 1 2
    The N extra samples are obtained by repeating the last sample value of the input signal N times to ensure signal continuity.

Where This Node Can Run:

Desktop OS: Windows

FPGA: Not supported

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