FFmpeg
4.0
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#include <libavcodec/acelp_filters.h>
Data Fields | |
void(* | acelp_interpolatef )(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length) |
Floating point version of ff_acelp_interpolate() More... | |
void(* | acelp_apply_order_2_transfer_function )(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n) |
Apply an order 2 rational transfer function in-place. More... | |
Definition at line 28 of file acelp_filters.h.
void(* ACELPFContext::acelp_interpolatef) (float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length) |
Floating point version of ff_acelp_interpolate()
Definition at line 32 of file acelp_filters.h.
Referenced by decode_pitch_vector(), ff_acelp_filter_init(), and ff_acelp_filter_init_mips().
void(* ACELPFContext::acelp_apply_order_2_transfer_function) (float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n) |
Apply an order 2 rational transfer function in-place.
out | output buffer for filtered speech samples |
in | input buffer containing speech data (may be the same as out) |
zero_coeffs | z^-1 and z^-2 coefficients of the numerator |
pole_coeffs | z^-1 and z^-2 coefficients of the denominator |
gain | scale factor for final output |
mem | intermediate values used by filter (should be 0 initially) |
n | number of samples (should be a multiple of eight) |
Definition at line 47 of file acelp_filters.h.
Referenced by amrnb_decode_frame(), amrwb_decode_frame(), ff_acelp_filter_init(), and ff_acelp_filter_init_mips().