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FFmpeg/libavcodec/celp_filters.h
Michael Niedermayer 686959e87e Merge remote-tracking branch 'qatar/master'
* qatar/master:
  doxygen: Consistently use '@' instead of '\' for Doxygen markup.
  Use av_printf_format to check the usage of printf style functions
  Add av_printf_format, for marking printf style format strings and their parameters
  ARM: enable thumb for Cortex-M* CPUs
  nsvdec: Propagate error values instead of returning 0 in nsv_read_header().
  build: remove SRC_PATH_BARE variable
  build: move basic rules and variables to main Makefile
  build: move special targets to end of main Makefile
  lavdev: improve feedback in case of invalid frame rate/size
  vfwcap: prefer "framerate_q" over "fps" in vfw_read_header()
  v4l2: prefer "framerate_q" over "fps" in v4l2_set_parameters()
  fbdev: prefer "framerate_q" over "fps" in device context
  bktr: prefer "framerate" over "fps" for grab_read_header()
  ALSA: implement channel layout for playback.
  alsa: support unsigned variants of already supported signed formats.
  alsa: add support for more formats.
  ARM: allow building in Thumb2 mode

Conflicts:
	common.mak
	doc/APIchanges
	libavcodec/vdpau.h
	libavdevice/alsa-audio-common.c
	libavdevice/fbdev.c
	libavdevice/libdc1394.c
	libavutil/avutil.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-06-24 03:07:04 +02:00

120 lines
4.4 KiB
C

/*
* various filters for CELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_CELP_FILTERS_H
#define AVCODEC_CELP_FILTERS_H
#include <stdint.h>
/**
* Circularly convolve fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
* @param fc_out vector with filter applied
* @param fc_in source vector
* @param 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!
*/
void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
const int16_t *filter, int len);
/**
* Add an array to a rotated array.
*
* out[k] = in[k] + fac * lagged[k-lag] with wrap-around
*
* @param out result vector
* @param in samples to be added unfiltered
* @param lagged samples to be rotated, multiplied and added
* @param lag lagged vector delay in the range [0, n]
* @param fac scalefactor for lagged samples
* @param n number of samples
*/
void ff_celp_circ_addf(float *out, const float *in,
const float *lagged, int lag, float fac, int n);
/**
* LP synthesis filter.
* @param[out] out pointer to output buffer
* @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
* @param in input signal
* @param buffer_length amount of data to process
* @param filter_length filter length (10 for 10th order LP filter)
* @param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
* @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
*
* @return 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.
*/
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.
* @param[out] out pointer to output buffer
* - the array out[-filter_length, -1] must
* contain the previous result of this filter
* @param filter_coeffs filter coefficients.
* @param in input signal
* @param buffer_length amount of data to process
* @param 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.
*/
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
const float *in, int buffer_length,
int filter_length);
/**
* LP zero synthesis filter.
* @param[out] out pointer to output buffer
* @param filter_coeffs filter coefficients.
* @param in input signal
* - the array in[-filter_length, -1] must
* contain the previous input of this filter
* @param buffer_length amount of data to process
* @param 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.
*/
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
const float *in, int buffer_length,
int filter_length);
#endif /* AVCODEC_CELP_FILTERS_H */