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FFmpeg/libavcodec/lsp.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

131 lines
4.5 KiB
C

/*
* LSP computing 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_LSP_H
#define AVCODEC_LSP_H
#include <stdint.h>
/**
(I.F) means fixed-point value with F fractional and I integer bits
*/
/**
* @brief ensure a minimum distance between LSFs
* @param[in,out] lsfq LSF to check and adjust
* @param lsfq_min_distance minimum distance between LSFs
* @param lsfq_min minimum allowed LSF value
* @param lsfq_max maximum allowed LSF value
* @param lp_order LP filter order
*/
void ff_acelp_reorder_lsf(int16_t* lsfq, int lsfq_min_distance, int lsfq_min, int lsfq_max, int lp_order);
/**
* Adjust the quantized LSFs so they are increasing and not too close.
*
* This step is not mentioned in the AMR spec but is in the reference C decoder.
* Omitting this step creates audible distortion on the sinusoidal sweep
* test vectors in 3GPP TS 26.074.
*
* @param[in,out] lsf LSFs in Hertz
* @param min_spacing minimum distance between two consecutive lsf values
* @param size size of the lsf vector
*/
void ff_set_min_dist_lsf(float *lsf, double min_spacing, int size);
/**
* @brief Convert LSF to LSP
* @param[out] lsp LSP coefficients (-0x8000 <= (0.15) < 0x8000)
* @param lsf normalized LSF coefficients (0 <= (2.13) < 0x2000 * PI)
* @param lp_order LP filter order
*
* @remark It is safe to pass the same array into the lsf and lsp parameters.
*/
void ff_acelp_lsf2lsp(int16_t *lsp, const int16_t *lsf, int lp_order);
/**
* Floating point version of ff_acelp_lsf2lsp()
*/
void ff_acelp_lsf2lspd(double *lsp, const float *lsf, int lp_order);
/**
* @brief LSP to LP conversion (3.2.6 of G.729)
* @param[out] lp decoded LP coefficients (-0x8000 <= (3.12) < 0x8000)
* @param lsp LSP coefficients (-0x8000 <= (0.15) < 0x8000)
* @param lp_half_order LP filter order, divided by 2
*/
void ff_acelp_lsp2lpc(int16_t* lp, const int16_t* lsp, int lp_half_order);
/**
* LSP to LP conversion (5.2.4 of AMR-WB)
*/
void ff_amrwb_lsp2lpc(const double *lsp, float *lp, int lp_order);
/**
* @brief Interpolate LSP for the first subframe and convert LSP -> LP for both subframes (3.2.5 and 3.2.6 of G.729)
* @param[out] lp_1st decoded LP coefficients for first subframe (-0x8000 <= (3.12) < 0x8000)
* @param[out] lp_2nd decoded LP coefficients for second subframe (-0x8000 <= (3.12) < 0x8000)
* @param lsp_2nd LSP coefficients of the second subframe (-0x8000 <= (0.15) < 0x8000)
* @param lsp_prev LSP coefficients from the second subframe of the previous frame (-0x8000 <= (0.15) < 0x8000)
* @param lp_order LP filter order
*/
void ff_acelp_lp_decode(int16_t* lp_1st, int16_t* lp_2nd, const int16_t* lsp_2nd, const int16_t* lsp_prev, int lp_order);
#define MAX_LP_HALF_ORDER 8
#define MAX_LP_ORDER (2*MAX_LP_HALF_ORDER)
/**
* Reconstruct LPC coefficients from the line spectral pair frequencies.
*
* @param lsp line spectral pairs in cosine domain
* @param lpc linear predictive coding coefficients
* @param lp_half_order half the number of the amount of LPCs to be
* reconstructed, need to be smaller or equal to MAX_LP_HALF_ORDER
*
* @note buffers should have a minimux size of 2*lp_half_order elements.
*
* TIA/EIA/IS-733 2.4.3.3.5
*/
void ff_acelp_lspd2lpc(const double *lsp, float *lpc, int lp_half_order);
/**
* Sort values in ascending order.
*
* @note O(n) if data already sorted, O(n^2) - otherwise
*/
void ff_sort_nearly_sorted_floats(float *vals, int len);
/**
* Compute the Pa / (1 + z(-1)) or Qa / (1 - z(-1)) coefficients
* needed for LSP to LPC conversion.
* We only need to calculate the 6 first elements of the polynomial.
*
* @param lsp line spectral pairs in cosine domain
* @param[out] f polynomial input/output as a vector
*
* TIA/EIA/IS-733 2.4.3.3.5-1/2
*/
void ff_lsp2polyf(const double *lsp, double *f, int lp_half_order);
#endif /* AVCODEC_LSP_H */