#include <stdint.h>
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Functions | |
void | ff_acelp_interpolate (int16_t *out, const int16_t *in, const int16_t *filter_coeffs, int precision, int frac_pos, int filter_length, int length) |
Generic FIR interpolation routine. | |
void | ff_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(). | |
void | ff_acelp_high_pass_filter (int16_t *out, int hpf_f[2], const int16_t *in, int length) |
high-pass filtering and upscaling (4.2.5 of G.729). | |
void | ff_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. | |
void | ff_tilt_compensation (float *mem, float tilt, float *samples, int size) |
Apply tilt compensation filter, 1 - tilt * z-1. | |
Variables | |
const int16_t | ff_acelp_interp_filter [61] |
low-pass Finite Impulse Response filter coefficients. |
void ff_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 |
Definition at line 117 of file acelp_filters.c.
Referenced by amrnb_decode_frame(), amrwb_decode_frame(), decode_frame(), and postfilter().
high-pass filtering and upscaling (4.2.5 of G.729).
[out] | out | output buffer for filtered speech data |
[in,out] | hpf_f | past filtered data from previous (2 items long) frames (-0x20000000 <= (14.13) < 0x20000000) |
in | speech data to process | |
length | input data size |
The filter has a cut-off frequency of 1/80 of the sampling freq
AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, but constants differs in 5th sign after comma). Fortunately in fixed-point all coefficients are the same as in G.729. Thus this routine can be used for the fixed-point AMR decoder, too.
Definition at line 97 of file acelp_filters.c.
Referenced by decode_frame().
void ff_acelp_interpolate | ( | int16_t * | out, | |
const int16_t * | in, | |||
const int16_t * | filter_coeffs, | |||
int | precision, | |||
int | frac_pos, | |||
int | filter_length, | |||
int | length | |||
) |
Generic FIR interpolation routine.
[out] | out | buffer for interpolated data |
in | input data | |
filter_coeffs | interpolation filter coefficients (0.15) | |
precision | sub sample factor, that is the precision of the position | |
frac_pos | fractional part of position [0..precision-1] | |
filter_length | filter length | |
length | length of output |
Definition at line 42 of file acelp_filters.c.
Referenced by decode_frame(), and long_term_filter().
void ff_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 76 of file acelp_filters.c.
Referenced by decode_frame(), decode_pitch_vector(), ff_sipr_decode_frame_16k(), and synth_block_fcb_acb().
void ff_tilt_compensation | ( | float * | mem, | |
float | tilt, | |||
float * | samples, | |||
int | size | |||
) |
Apply tilt compensation filter, 1 - tilt * z-1.
mem | pointer to the filter's state (one single float) | |
tilt | tilt factor | |
samples | array where the filter is applied | |
size | the size of the samples array |
Definition at line 134 of file acelp_filters.c.
Referenced by calc_input_response(), postfilter(), postfilter_5k0(), and wiener_denoise().
const int16_t ff_acelp_interp_filter[61] |
low-pass Finite Impulse Response filter coefficients.
Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq, the coefficients are scaled by 2^15. This array only contains the right half of the filter. This filter is likely identical to the one used in G.729, though this could not be determined from the original comments with certainity.
Definition at line 28 of file acelp_filters.c.
Referenced by decode_frame().