Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

1 files changed, 301 insertions, 0 deletions

diff --git a/fdk-aac/libFDK/include/qmf.h b/fdk-aac/libFDK/include/qmf.h
new file mode 100644
index 0000000..609c6f1
--- /dev/null
+++ b/fdk-aac/libFDK/include/qmf.h
@@ -0,0 +1,301 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 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
+----------------------------------------------------------------------------- */
+
+/******************* Library for basic calculation routines ********************
+
+   Author(s):
+
+   Description:
+
+*******************************************************************************/
+
+/*!
+  \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"
+
+#define FIXP_QAS FIXP_PCM
+#define QAS_BITS SAMPLE_BITS
+
+#define FIXP_QSS FIXP_DBL
+#define QSS_BITS DFRACT_BITS
+
+/* Flags for QMF intialization */
+/* Low Power mode flag */
+#define QMF_FLAG_LP 1
+/* Filter is not symmetric. This flag is set internally in the QMF
+ * initialization as required. */
+/* DO NOT PASS THIS FLAG TO qmfInitAnalysisFilterBank or
+ * qmfInitSynthesisFilterBank */
+#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
+
+#define QMF_MAX_SYNTHESIS_BANDS (64)
+
+/*!
+ * \brief Algorithmic scaling in sbrForwardModulation()
+ *
+ * The scaling in sbrForwardModulation() is caused by:
+ *
+ *   \li 1 R_SHIFT in sbrForwardModulation()
+ *   \li 5/6 R_SHIFT in dct3() if using 32/64 Bands
+ *   \li 1 omitted gain of 2.0 in qmfForwardModulation()
+ */
+#define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7
+
+/*!
+ * \brief Algorithmic scaling in cplxSynthesisQmfFiltering()
+ *
+ * The scaling in cplxSynthesisQmfFiltering() is caused by:
+ *
+ *   \li  5/6 R_SHIFT in dct2() if using 32/64 Bands
+ *   \li  1 omitted gain of 2.0 in qmfInverseModulation()
+ *   \li -6 division by 64 in synthesis filterbank
+ *   \li x bits external influence
+ */
+#define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1
+
+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 synScalefactor; /*!< Scale factor of synthesis qmf (syn only) */
+  int outScalefactor; /*!< Scale factor of output data (syn only) */
+  FIXP_DBL outGain_m; /*!< Mantissa of gain output data (syn only) (init with
+                         0x80000000 to ignore) */
+  int outGain_e;      /*!< Exponent of gain output data (syn only) */
+
+  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_DBL **qmfReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
+    QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
+    const LONG *timeIn,            /*!< Time signal */
+    const int timeIn_e,            /*!< Exponent of audio data        */
+    const int stride,              /*!< Stride factor of audio data   */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
+);
+
+void qmfAnalysisFiltering(
+    HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank   */
+    FIXP_DBL **qmfReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **qmfImag,            /*!< Pointer to imag subband slots */
+    QMF_SCALE_FACTOR *scaleFactor, /*!< Scale factors of QMF data     */
+    const INT_PCM *timeIn,         /*!< Time signal */
+    const int timeIn_e,            /*!< Exponent of audio data        */
+    const int stride,              /*!< Stride factor of audio data   */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporal working buffer */
+);
+
+void qmfSynthesisFiltering(
+    HANDLE_QMF_FILTER_BANK synQmf,       /*!< Handle of Qmf Synthesis Bank  */
+    FIXP_DBL **QmfBufferReal,            /*!< Pointer to real subband slots */
+    FIXP_DBL **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_DBL *pWorkBuffer /*!< pointer to temporary working buffer. It must be
+                             aligned */
+);
+
+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_DBL *qmfReal,             /*!< Low and High band, real */
+    FIXP_DBL *qmfImag,             /*!< Low and High band, imag */
+    const LONG *timeIn,            /*!< Pointer to input */
+    const int stride,              /*!< stride factor of input */
+    FIXP_DBL *pWorkBuffer          /*!< pointer to temporary working buffer */
+);
+
+void qmfAnalysisFilteringSlot(
+    HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank  */
+    FIXP_DBL *qmfReal,             /*!< Low and High band, real */
+    FIXP_DBL *qmfImag,             /*!< Low and High band, imag */
+    const INT_PCM *timeIn,         /*!< Pointer to input */
+    const int stride,              /*!< stride factor of input */
+    FIXP_DBL *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_DBL *realSlot,
+                               const FIXP_DBL *imagSlot,
+                               const int scaleFactorLowBand,
+                               const int scaleFactorHighBand, INT_PCM *timeOut,
+                               const int timeOut_e, FIXP_DBL *pWorkBuffer);
+
+void qmfChangeOutScalefactor(
+    HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
+    int outScalefactor             /*!< New scaling factor for output data */
+);
+
+int qmfGetOutScalefactor(
+    HANDLE_QMF_FILTER_BANK synQmf /*!< Handle of Qmf Synthesis Bank */
+);
+
+void qmfChangeOutGain(
+    HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */
+    FIXP_DBL outputGain,           /*!< New gain for output data (mantissa) */
+    int outputGainScale            /*!< New gain for output data (exponent) */
+);
+void qmfSynPrototypeFirSlot(
+    HANDLE_QMF_FILTER_BANK qmf,
+    FIXP_DBL *RESTRICT realSlot, /*!< Input: Pointer to real Slot */
+    FIXP_DBL *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */
+    INT_PCM *RESTRICT timeOut,   /*!< Time domain data */
+    const int timeOut_e);
+
+#endif /*ifndef QMF_H       */