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authorThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
committerThe Android Open Source Project <initial-contribution@android.com>2012-07-11 10:15:24 -0700
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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 */