diff --git a/audio/filters/EQ.dsp b/audio/filters/EQ.dsp new file mode 100644 index 0000000000..68d733f3fc --- /dev/null +++ b/audio/filters/EQ.dsp @@ -0,0 +1,29 @@ +filters = 1 +filter0 = eq + +# Defaults + +# Beta factor for Kaiser window. +# Lower values will allow better frequency resolution, but more ripple. +# eq_window_beta = 4.0 + +# The block size on which FFT is done. +# Too high value requires more processing as well as longer latency but +# allows finer-grained control over the spectrum. +# eq_block_size_log2 = 8 + +# An array of which frequencies to control. +# You can create an arbitrary amount of these sampling points. +# The EQ will try to create a frequency response which fits well to these points. +# The filter response is linearly interpolated between sampling points here. +# +# It is implied that 0 Hz (DC) and Nyquist have predefined gains of 0 dB which are interpolated against. +# If you want a "peak" in the spectrum or similar, you have to define close points to say, 0 dB. +# +# E.g.: A boost of 3 dB at 1 kHz can be expressed as. +# eq_frequencies = "500 1000 2000" +# eq_gains = "0 3 0" +# Due to frequency domain smearing, you will not get exactly +3 dB at 1 kHz. + +# By default, this filter has a flat frequency response. +