libavcodec/celp_filters.c File Reference

#include <inttypes.h>
#include "avcodec.h"
#include "celp_filters.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 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.

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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.

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	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
	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 ff_subblock_synthesis().

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 196 of file celp_filters.c.

Referenced by postfilter(), and postfilter_5k0().