libavcodec/celp_filters.h File Reference

#include <stdint.h>

Go to the source code of this file.

Functions

void ff_celp_convolve_circ (int16_t *fc_out, const int16_t *fc_in, const int16_t *filter, int len)
 Circularly convolve fixed vector with a phase dispersion impulse response filter (D.6.2 of G.729 and 6.1.5 of AMR).
void ff_celp_circ_addf (float *out, const float *in, const float *lagged, int lag, float fac, int n)
 Add an array to a rotated array.
int ff_celp_lp_synthesis_filter (int16_t *out, const int16_t *filter_coeffs, const int16_t *in, int buffer_length, int filter_length, int stop_on_overflow, int shift, int rounder)
 LP synthesis filter.
void ff_celp_lp_synthesis_filterf (float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
 LP synthesis filter.
void ff_celp_lp_zero_synthesis_filterf (float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
 LP zero synthesis filter.


Function Documentation

void ff_celp_circ_addf ( float *  out,
const float *  in,
const float *  lagged,
int  lag,
float  fac,
int  n 
)

Add an array to a rotated array.

out[k] = in[k] + fac * lagged[k-lag] with wrap-around

Parameters:
out result vector
in samples to be added unfiltered
lagged samples to be rotated, multiplied and added
lag lagged vector delay in the range [0, n]
fac scalefactor for lagged samples
n number of samples

Definition at line 48 of file celp_filters.c.

Referenced by anti_sparseness(), and apply_ir_filter().

void ff_celp_convolve_circ ( int16_t fc_out,
const int16_t fc_in,
const int16_t filter,
int  len 
)

Circularly convolve fixed vector with a phase dispersion impulse response filter (D.6.2 of G.729 and 6.1.5 of AMR).

Parameters:
fc_out vector with filter applied
fc_in source vector
filter phase filter coefficients
fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)len] }

Note:
fc_in and fc_out should not overlap!

Definition at line 28 of file celp_filters.c.

Referenced by g729d_get_new_exc().

int ff_celp_lp_synthesis_filter ( int16_t out,
const int16_t filter_coeffs,
const int16_t in,
int  buffer_length,
int  filter_length,
int  stop_on_overflow,
int  shift,
int  rounder 
)

LP synthesis filter.

Parameters:
[out] out pointer to output buffer
filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
in input signal
buffer_length amount of data to process
filter_length filter length (10 for 10th order LP filter)
stop_on_overflow 1 - return immediately if overflow occurs 0 - ignore overflows
shift the result is shifted right by this value
rounder the amount to add for rounding (usually 0x800 or 0xfff)
Returns:
1 if overflow occurred, 0 - otherwise
Note:
Output buffer must contain filter_length samples of past speech data before pointer.
Routine applies 1/A(z) filter to given speech data.

Definition at line 58 of file celp_filters.c.

Referenced by decode_frame(), ff_g729_postfilter(), ff_subblock_synthesis(), g723_1_decode_frame(), and get_tilt_comp().

void ff_celp_lp_synthesis_filterf ( float *  out,
const float *  filter_coeffs,
const float *  in,
int  buffer_length,
int  filter_length 
)

LP synthesis filter.

Parameters:
[out] out pointer to output buffer
  • the array out[-filter_length, -1] must contain the previous result of this filter
filter_coeffs filter coefficients.
in input signal
buffer_length amount of data to process
filter_length filter length (10 for 10th order LP filter). Must be greater than 4 and even.
Note:
Output buffer must contain filter_length samples of past speech data before pointer.
Routine applies 1/A(z) filter to given speech data.

Definition at line 83 of file celp_filters.c.

Referenced by adaptive_cb_search(), decode(), decode_frame(), eval_ir(), ff_sipr_decode_frame_16k(), fixed_cb_search(), get_match_score(), hb_synthesis(), postfilter(), postfilter_5k0(), qcelp_decode_frame(), ra144_encode_subblock(), synth_block(), synthesis(), and tilt_factor().

void ff_celp_lp_zero_synthesis_filterf ( float *  out,
const float *  filter_coeffs,
const float *  in,
int  buffer_length,
int  filter_length 
)

LP zero synthesis filter.

Parameters:
[out] out pointer to output buffer
filter_coeffs filter coefficients.
in input signal
  • the array in[-filter_length, -1] must contain the previous input of this filter
buffer_length amount of data to process
filter_length filter length (10 for 10th order LP filter)
Note:
Output buffer must contain filter_length samples of past speech data before pointer.
Routine applies A(z) filter to given speech data.

Definition at line 198 of file celp_filters.c.

Referenced by postfilter(), and postfilter_5k0().


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