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, 281 insertions, 0 deletions

diff --git a/fdk-aac/libAACdec/src/usacdec_acelp.h b/fdk-aac/libAACdec/src/usacdec_acelp.h
new file mode 100644
index 0000000..9de41ff
--- /dev/null
+++ b/fdk-aac/libAACdec/src/usacdec_acelp.h
@@ -0,0 +1,281 @@
+/* -----------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------- */
+
+/**************************** AAC decoder library ******************************
+
+   Author(s):   Matthias Hildenbrand
+
+   Description: USAC ACELP frame decoder
+
+*******************************************************************************/
+
+#ifndef USACDEC_ACELP_H
+#define USACDEC_ACELP_H
+
+#include "common_fix.h"
+#include "FDK_bitstream.h"
+#include "usacdec_const.h"
+#include "usacdec_rom.h"
+
+//#define ENHANCED_TCX_TD_CONCEAL_ENABLE
+
+/** Structure which holds the ACELP internal persistent memory */
+typedef struct {
+  FIXP_DBL old_exc_mem[PIT_MAX_MAX + L_INTERPOL];
+  FIXP_DBL old_syn_mem[M_LP_FILTER_ORDER]; /* synthesis filter states */
+  FIXP_SGL A[M_LP_FILTER_ORDER];
+  INT A_exp;
+  FIXP_DBL gc_threshold;
+  FIXP_DBL de_emph_mem;
+  FIXP_SGL past_gpit;
+  FIXP_DBL past_gcode;
+  USHORT old_T0;
+  UCHAR old_T0_frac;
+  FIXP_DBL deemph_mem_wsyn;
+  FIXP_DBL wsyn_rms;
+  SHORT seed_ace;
+} CAcelpStaticMem;
+
+/** Structure which holds the parameter data needed to decode one ACELP frame.
+ */
+typedef struct {
+  UCHAR
+  acelp_core_mode;   /**< mean excitation energy index for whole ACELP frame
+                      */
+  UCHAR mean_energy; /**< acelp core mode for whole ACELP frame */
+  USHORT T0[NB_SUBFR];
+  UCHAR T0_frac[NB_SUBFR];
+  UCHAR ltp_filtering_flag[NB_SUBFR]; /**< controlls whether LTP postfilter is
+                                         active for each ACELP subframe */
+  SHORT icb_index[NB_SUBFR]
+                 [8]; /**< innovative codebook index for each ACELP subframe */
+  UCHAR gains[NB_SUBFR]; /**< gain index for each ACELP subframe */
+} CAcelpChannelData;
+
+/**
+ * \brief Read the acelp_coding() bitstream part.
+ * \param[in] hBs bitstream handle to read data from.
+ * \param[out] acelpData pointer to structure to store the parsed data of one
+ * ACELP frame.
+ * \param[in] acelp_core_mode the ACELP core mode index.
+ * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
+ */
+INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelpData,
+                   INT acelp_core_mode, INT i_offset, INT coreCoderFrameLength);
+/**
+ * \brief Initialization of memory before one LPD frame is decoded
+ * \param[out] synth_buf synthesis buffer to be initialized, exponent = SF_SYNTH
+ * \param[in] old_synth past synthesis of previous LPD frame, exponent =
+ * SF_SYNTH
+ * \param[out] synth_buf_fb fullband synthesis buffer to be initialized,
+ * exponent = SF_SYNTH
+ * \param[in] old_synth_fb past fullband synthesis of previous LPD frame,
+ * exponent = SF_SYNTH
+ * \param[out] pitch vector where decoded pitch lag values are stored
+ * \param[in] old_T_pf past pitch lag values of previous LPD frame
+ * \param[in] samplingRate sampling rate for pitch lag offset calculation
+ * \param[out] i_offset pitch lag offset for the decoding of the pitch lag
+ * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
+ */
+void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch,
+                         INT *old_T_pf, FIXP_DBL *pit_gain,
+                         FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset,
+                         INT coreCoderFrameLength, INT synSfd,
+                         INT nbSubfrSuperfr);
+
+/**
+ * \brief Save tail of buffers for the initialization of the next LPD frame
+ * \param[in] synth_buf synthesis of current LPD frame, exponent = SF_SYNTH
+ * \param[out] old_synth memory where tail of fullband synth_buf is stored,
+ * exponent = SF_SYNTH
+ * \param[in] synth_buf_fb fullband synthesis of current LPD frame, exponent =
+ * SF_SYNTH
+ * \param[out] old_synth_fb memory where tail of fullband synth_buf is stored,
+ * exponent = SF_SYNTH
+ * \param[in] pitch decoded pitch lag values of current LPD frame
+ * \param[out] old_T_pf memory where last SYN_SFD pitch lag values are stored
+ */
+void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch,
+                          INT *old_T_pf, INT coreCoderFrameLength, INT synSfd,
+                          INT nbSubfrSuperfr);
+
+/**
+ * \brief Decode one ACELP frame (three or four ACELP subframes with 64 samples
+ * each)
+ * \param[in,out] acelp_mem pointer to ACELP memory structure
+ * \param[in] i_offset pitch lag offset
+ * \param[in] lsp_old LPC filter in LSP domain corresponding to previous frame
+ * \param[in] lsp_new LPC filter in LSP domain corresponding to current frame
+ * \param[in] stab_fac stability factor constrained by 0<=stab_fac<=1.0,
+ * exponent = SF_STAB
+ * \param[in] acelpData pointer to struct with data which is needed for decoding
+ * one ACELP frame
+ * \param[out] synth ACELP output signal
+ * \param[out] pT four decoded pitch lag values
+ * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
+ */
+void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset,
+                      const FIXP_LPC lsp_old[M_LP_FILTER_ORDER],
+                      const FIXP_LPC lsp_new[M_LP_FILTER_ORDER],
+                      FIXP_SGL stab_fac, CAcelpChannelData *acelpData,
+                      INT numLostSubframes, int lastLpcLost, int frameCnt,
+                      FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain,
+                      INT coreCoderFrameLength);
+
+/**
+ * \brief Reset ACELP internal memory.
+ * \param[out] acelp_mem pointer to ACELP memory structure
+ */
+void CLpd_AcelpReset(CAcelpStaticMem *acelp_mem);
+
+/**
+ * \brief Initialize ACELP internal memory in case of FAC before ACELP decoder
+ * is called
+ * \param[in] synth points to end+1 of past valid synthesis signal, exponent =
+ * SF_SYNTH
+ * \param[in] last_lpd_mode last lpd mode
+ * \param[in] last_last_lpd_mode lpd mode before last_lpd_mode
+ * \param[in] A_new LP synthesis filter coeffs corresponding to last frame,
+ * exponent = SF_A_COEFFS
+ * \param[in] A_old LP synthesis filter coeffs corresponding to the frame before
+ * last frame, exponent = SF_A_COEFFS
+ * \param[in,out] acelp_mem pointer to ACELP memory structure
+ * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
+ */
+void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode,
+                                  UCHAR last_last_lpd_mode,
+                                  const FIXP_LPC *A_new, const INT A_new_exp,
+                                  const FIXP_LPC *A_old, const INT A_old_exp,
+                                  CAcelpStaticMem *acelp_mem,
+                                  INT coreCoderFrameLength, INT clearOldExc,
+                                  UCHAR lpd_mode);
+
+/**
+ * \brief Calculate zero input response (zir) of the acelp synthesis filter
+ * \param[in] A LP synthesis filter coefficients, exponent = SF_A_COEFFS
+ * \param[in,out] acelp_mem pointer to ACELP memory structure
+ * \param[in] length length of zir
+ * \param[out] zir pointer to zir output buffer, exponent = SF_SYNTH
+ */
+void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp,
+                    CAcelpStaticMem *acelp_mem, const INT length,
+                    FIXP_DBL zir[], int doDeemph);
+
+/**
+ * \brief Borrow static excitation memory from ACELP decoder
+ * \param[in] acelp_mem pointer to ACELP memory structure
+ * \param[in] length number of requested FIXP_DBL values
+ * \return pointer to requested memory
+ *
+ * The caller has to take care not to overwrite valid memory areas.
+ * During TCX/FAC calculations and before CLpd_AcelpPrepareInternalMem() is
+ * called, the following memory size is available:
+ * - 256 samples in case of ACELP -> TCX20 -> ACELP transition
+ * - PIT_MAX_MAX+L_INTERPOL samples in all other cases
+ */
+FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length);
+
+void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch,
+                       const FIXP_LPC lsp_old[M_LP_FILTER_ORDER],
+                       const FIXP_LPC lsp_new[M_LP_FILTER_ORDER],
+                       const FIXP_SGL stab_fac, INT numLostSubframes,
+                       FIXP_DBL synth[], INT coreCoderFrameLength,
+                       UCHAR last_tcx_noise_factor);
+
+inline SHORT E_UTIL_random(SHORT *seed) {
+  *seed = (SHORT)((((LONG)*seed * (LONG)31821) >> 1) + (LONG)13849);
+  return (*seed);
+}
+
+#endif /* USACDEC_ACELP_H */