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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+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
+
+
+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 */