***MUST USE MATLAB***

-Analog-to-digital Filter Transformation: Impulse Invariance method.

An analog RC-low-pass filter has cut-off frequency set at 100Hz. Use impulse invariance method to transform the filter to digital domain with sampling rate of 2000 sps.

Digital filter coefficients:

b = _________

a = _________

Insert the amplitude response of the digital system and compare it with the original analog filter.

-Analog-to-digital Filter Transformation: Bilinear Transformation method:

An analog RC-low-pass filter has cut-off frequency set at 100Hz. Use bilinear transformation method to transform the filter to digital domain with sampling rate of 2000 sps

Digital filter coefficients:

b = _________

a = _________

Insert the amplitude response of the digital system and compare it with the original analog filter and Impulse Invariance method design.

-Design an IIR low pass filter with the following properties:

sampling rate =22050

cutoff frequency = 700Hz

stop-band attenuation more than 55dB

transition bandwidth = 350Hz

Max 2dB ripple on passband

Minimum order number

Verify your design from amplitude response graph.

Save the filter coefficients to C-language .h-header file.

Insert the amplitude response figure here!

Link to filter coefficients .h-file.

-Design IIR-filters for a 3-band equalizer:

Sampling rate = 44100

Bass = 0-200 Hz

Middle = 200 Hz - 2000 Hz

Treble = 2000 Hz - Fn

Stopband attenuation 40dB

Flat (no ripple) passband

Transition bandwidths = fc/4

Verify your design from the amplitude response graphs.

Write commands to generate filter coefficients for each band.

Insert the amplitude response figures here.

-Combine the previous filters to adjust the tone of a voice sample:

Bass: +10db

Middle: -15dB

Treble: +12dB

You may choose your own sound sample freely.

Write the commands (and provide links to .fda-files if using filterDesigner)

Provide links to both original voice sample and the filtered one.