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-
-/* -----------------------------------------------------------------------------------------------------------
-Software License for 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 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 */