libavcodec/acelp_filters.h - nest-android-app/ffmpeg - Git at Google

 /*
  * various filters for ACELP-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_ACELP_FILTERS_H
 #define AVCODEC_ACELP_FILTERS_H

 #include <stdint.h>

 typedef struct ACELPFContext {
     /**
     * Floating point version of ff_acelp_interpolate()
     */
     void (*acelp_interpolatef)(float *out, const float *in,
                             const float *filter_coeffs, int precision,
                             int frac_pos, int filter_length, int length);

     /**
      * Apply an order 2 rational transfer function in-place.
      *
      * @param out output buffer for filtered speech samples
      * @param in input buffer containing speech data (may be the same as out)
      * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
      * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
      * @param gain scale factor for final output
      * @param mem intermediate values used by filter (should be 0 initially)
      * @param n number of samples (should be a multiple of eight)
      */
     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);

 }ACELPFContext;

 /**
  * Initialize ACELPFContext.
  */
 void ff_acelp_filter_init(ACELPFContext *c);
 void ff_acelp_filter_init_mips(ACELPFContext *c);

 /**
  * 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 certainty.
  */
 extern const int16_t ff_acelp_interp_filter[61];

 /**
  * Generic FIR interpolation routine.
  * @param[out] out buffer for interpolated data
  * @param in input data
  * @param filter_coeffs interpolation filter coefficients (0.15)
  * @param precision sub sample factor, that is the precision of the position
  * @param frac_pos fractional part of position [0..precision-1]
  * @param filter_length filter length
  * @param length length of output
  *
  * filter_coeffs contains coefficients of the right half of the symmetric
  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
  * See ff_acelp_interp_filter for an example.
  *
  */
 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);

 /**
  * Floating point version of ff_acelp_interpolate()
  */
 void ff_acelp_interpolatef(float *out, const float *in,
                            const float *filter_coeffs, int precision,
                            int frac_pos, int filter_length, int length);


 /**
  * high-pass filtering and upscaling (4.2.5 of G.729).
  * @param[out]     out   output buffer for filtered speech data
  * @param[in,out]  hpf_f past filtered data from previous (2 items long)
  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
  * @param in speech data to process
  * @param length input data size
  *
  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
  *
  * The filter has a cut-off frequency of 1/80 of the sampling freq
  *
  * @note Two items before the top of the in buffer must contain two items from the
  *       tail of the previous subframe.
  *
  * @remark It is safe to pass the same array in in and out parameters.
  *
  * @remark 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.
  */
 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
                                const int16_t* in, int length);

 /**
  * Apply an order 2 rational transfer function in-place.
  *
  * @param out output buffer for filtered speech samples
  * @param in input buffer containing speech data (may be the same as out)
  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
  * @param gain scale factor for final output
  * @param mem intermediate values used by filter (should be 0 initially)
  * @param n number of samples
  */
 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 tilt compensation filter, 1 - tilt * z-1.
  *
  * @param mem pointer to the filter's state (one single float)
  * @param tilt tilt factor
  * @param samples array where the filter is applied
  * @param size the size of the samples array
  */
 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);


 #endif /* AVCODEC_ACELP_FILTERS_H */
	/*
	* various filters for ACELP-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_ACELP_FILTERS_H
	#define AVCODEC_ACELP_FILTERS_H

	#include <stdint.h>

	typedef struct ACELPFContext {
	/**
	* Floating point version of ff_acelp_interpolate()
	*/
	void (acelp_interpolatef)(float out, const float *in,
	const float *filter_coeffs, int precision,
	int frac_pos, int filter_length, int length);

	/**
	* Apply an order 2 rational transfer function in-place.
	*
	* @param out output buffer for filtered speech samples
	* @param in input buffer containing speech data (may be the same as out)
	* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
	* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
	* @param gain scale factor for final output
	* @param mem intermediate values used by filter (should be 0 initially)
	* @param n number of samples (should be a multiple of eight)
	*/
	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);

	}ACELPFContext;

	/**
	* Initialize ACELPFContext.
	*/
	void ff_acelp_filter_init(ACELPFContext *c);
	void ff_acelp_filter_init_mips(ACELPFContext *c);

	/**
	* 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 certainty.
	*/
	extern const int16_t ff_acelp_interp_filter[61];

	/**
	* Generic FIR interpolation routine.
	* @param[out] out buffer for interpolated data
	* @param in input data
	* @param filter_coeffs interpolation filter coefficients (0.15)
	* @param precision sub sample factor, that is the precision of the position
	* @param frac_pos fractional part of position [0..precision-1]
	* @param filter_length filter length
	* @param length length of output
	*
	* filter_coeffs contains coefficients of the right half of the symmetric
	* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
	* See ff_acelp_interp_filter for an example.
	*
	*/
	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);

	/**
	* Floating point version of ff_acelp_interpolate()
	*/
	void ff_acelp_interpolatef(float out, const float in,
	const float *filter_coeffs, int precision,
	int frac_pos, int filter_length, int length);


	/**
	* high-pass filtering and upscaling (4.2.5 of G.729).
	* @param[out] out output buffer for filtered speech data
	* @param[in,out] hpf_f past filtered data from previous (2 items long)
	* frames (-0x20000000 <= (14.13) < 0x20000000)
	* @param in speech data to process
	* @param length input data size
	*
	* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
	* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
	*
	* The filter has a cut-off frequency of 1/80 of the sampling freq
	*
	* @note Two items before the top of the in buffer must contain two items from the
	* tail of the previous subframe.
	*
	* @remark It is safe to pass the same array in in and out parameters.
	*
	* @remark 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.
	*/
	void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
	const int16_t* in, int length);

	/**
	* Apply an order 2 rational transfer function in-place.
	*
	* @param out output buffer for filtered speech samples
	* @param in input buffer containing speech data (may be the same as out)
	* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
	* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
	* @param gain scale factor for final output
	* @param mem intermediate values used by filter (should be 0 initially)
	* @param n number of samples
	*/
	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 tilt compensation filter, 1 - tilt * z-1.
	*
	* @param mem pointer to the filter's state (one single float)
	* @param tilt tilt factor
	* @param samples array where the filter is applied
	* @param size the size of the samples array
	*/
	void ff_tilt_compensation(float mem, float tilt, float samples, int size);


	#endif /* AVCODEC_ACELP_FILTERS_H */