
/* -----------------------------------------------------------------------------------------------------------
Software License for The Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
  All rights reserved.

 1.    INTRODUCTION
The Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
This FDK AAC Codec software is intended to be used on a wide variety of Android devices.

AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
of the MPEG specifications.

Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
individually for the purpose of encoding or decoding bit streams in products that are compliant with
the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
software may already be covered under those patent licenses when it is used for those licensed purposes only.

Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
applications information and documentation.

2.    COPYRIGHT LICENSE

Redistribution and use in source and binary forms, with or without modification, are permitted without
payment of copyright license fees provided that you satisfy the following conditions:

You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
your modifications thereto in source code form.

You must retain the complete text of this software license in the documentation and/or other materials
provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.

The name of Fraunhofer may not be used to endorse or promote products derived from this library without
prior written permission.

You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
software or your modifications thereto.

Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
and the date of any change. For modified versions of the FDK AAC Codec, the term
"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."

3.    NO PATENT LICENSE

NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
respect to this software.

You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
by appropriate patent licenses.

4.    DISCLAIMER

This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
or business interruption, however caused and on any theory of liability, whether in contract, strict
liability, or tort (including negligence), arising in any way out of the use of this software, even if
advised of the possibility of such damage.

5.    CONTACT INFORMATION

Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany

www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */

/*!
  \file   qmf.h
  \brief  Complex qmf analysis/synthesis   
  \author Markus Werner

*/
#ifndef __QMF_H
#define __QMF_H



#include "common_fix.h"
#include "FDK_tools_rom.h"
#include "dct.h"

/*
 * Filter coefficient type definition
 */
#ifdef QMF_DATA_16BIT
#define FIXP_QMF FIXP_SGL
#define FX_DBL2FX_QMF FX_DBL2FX_SGL
#define FX_QMF2FX_DBL FX_SGL2FX_DBL
#define QFRACT_BITS FRACT_BITS
#else
#define FIXP_QMF FIXP_DBL
#define FX_DBL2FX_QMF
#define FX_QMF2FX_DBL
#define QFRACT_BITS DFRACT_BITS
#endif

/* ARM neon optimized QMF analysis filter requires 32 bit input.
   Implemented for RVCT only, currently disabled. See src/arm/qmf_arm.cpp:45 */
#define FIXP_QAS FIXP_PCM
#define QAS_BITS SAMPLE_BITS

#ifdef QMFSYN_STATES_16BIT
#define FIXP_QSS FIXP_SGL
#define QSS_BITS FRACT_BITS
#else
#define FIXP_QSS FIXP_DBL
#define QSS_BITS DFRACT_BITS
#endif

/* Flags for QMF intialization */
/* Low Power mode flag */
#define QMF_FLAG_LP           1
/* Filter is not symetric. This flag is set internally in the QMF initialization as required. */
#define QMF_FLAG_NONSYMMETRIC 2
/* Complex Low Delay Filter Bank (or std symmetric filter bank) */
#define QMF_FLAG_CLDFB        4
/* Flag indicating that the states should be kept. */
#define QMF_FLAG_KEEP_STATES  8
/* Complex Low Delay Filter Bank used in MPEG Surround Encoder */
#define QMF_FLAG_MPSLDFB     16
/* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */
#define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION  32
/* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */
#define QMF_FLAG_DOWNSAMPLED  64


typedef struct
{
  int lb_scale;        /*!< Scale of low band area                   */
  int ov_lb_scale;     /*!< Scale of adjusted overlap low band area  */
  int hb_scale;        /*!< Scale of high band area                  */
  int ov_hb_scale;     /*!< Scale of adjusted overlap high band area */
} QMF_SCALE_FACTOR;

struct QMF_FILTER_BANK
{
  const FIXP_PFT *p_filter;     /*!< Pointer to filter coefficients */

  void *FilterStates;           /*!< Pointer to buffer of filter states
                                     FIXP_PCM in analyse and
                                     FIXP_DBL in synthesis filter */
  int FilterSize;               /*!< Size of prototype filter. */
  const FIXP_QTW *t_cos;        /*!< Modulation tables. */
  const FIXP_QTW *t_sin;
  int filterScale;              /*!< filter scale */

  int no_channels;              /*!< Total number of channels (subbands) */
  int no_col;                   /*!< Number of time slots       */
  int lsb;                      /*!< Top of low subbands */
  int usb;                      /*!< Top of high subbands */

  int outScalefactor;           /*!< Scale factor of output data (syn only) */
  FIXP_DBL outGain;             /*!< Gain output data (syn only) (init with 0x80000000 to ignore) */

  UINT flags;                   /*!< flags */
  UCHAR p_stride;               /*!< Stride Factor of polyphase filters */

};

typedef struct QMF_FILTER_BANK *HANDLE_QMF_FILTER_BANK;

void
qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf,  /*!< Handle of Qmf Analysis Bank   */
                      FIXP_QMF **qmfReal,             /*!< Pointer to real subband slots */
                      FIXP_QMF **qmfImag,             /*!< Pointer to imag subband slots */
                      QMF_SCALE_FACTOR *scaleFactor,  /*!< Scale factors of QMF data     */
                      const INT_PCM *timeIn,          /*!< Time signal */
                      const int  stride,              /*!< Stride factor of audio data   */
                      FIXP_QMF  *pWorkBuffer          /*!< pointer to temporal working buffer */
                      );

void
qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
                       FIXP_QMF  **QmfBufferReal,           /*!< Pointer to real subband slots */
                       FIXP_QMF  **QmfBufferImag,           /*!< Pointer to imag subband slots */
                       const QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
                       const int   ov_len,                  /*!< Length of band overlap        */
                       INT_PCM    *timeOut,                 /*!< Time signal */
                       const int   stride,                  /*!< Stride factor of audio data   */
                       FIXP_QMF   *pWorkBuffer              /*!< pointer to temporal working buffer */
                       );

int
qmfInitAnalysisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
                           FIXP_QAS *pFilterStates,      /*!< Pointer to filter state buffer */
                           int noCols,                   /*!< Number of time slots  */
                           int lsb,                      /*!< Number of lower bands */
                           int usb,                      /*!< Number of upper bands */
                           int no_channels,              /*!< Number of critically sampled bands */
                           int flags);                   /*!< Flags */

void
qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf,  /*!< Handle of Qmf Synthesis Bank  */
                          FIXP_QMF      *qmfReal,         /*!< Low and High band, real */
                          FIXP_QMF      *qmfImag,         /*!< Low and High band, imag */
                          const INT_PCM *timeIn,          /*!< Pointer to input */
                          const int      stride,          /*!< stride factor of input */
                          FIXP_QMF      *pWorkBuffer      /*!< pointer to temporal working buffer */
                         );

int
qmfInitSynthesisFilterBank( HANDLE_QMF_FILTER_BANK h_Qmf, /*!< QMF Handle */
                            FIXP_QSS *pFilterStates,      /*!< Pointer to filter state buffer */
                            int noCols,                   /*!< Number of time slots  */
                            int lsb,                      /*!< Number of lower bands */
                            int usb,                      /*!< Number of upper bands */
                            int no_channels,              /*!< Number of critically sampled bands */
                            int flags);                   /*!< Flags */

void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK  synQmf,
                                const FIXP_QMF *realSlot,
                                const FIXP_QMF *imagSlot,
                                const int       scaleFactorLowBand,
                                const int       scaleFactorHighBand,
                                INT_PCM        *timeOut,
                                const int       stride,
                                FIXP_QMF       *pWorkBuffer);

void
qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf,     /*!< Handle of Qmf Synthesis Bank */
                         int outScalefactor                 /*!< New scaling factor for output data */
                        );

void
qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf,     /*!< Handle of Qmf Synthesis Bank */
                  FIXP_DBL outputGain                /*!< New gain for output data */
                 );



#endif /* __QMF_H */
