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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
commit2228e360595641dd906bf1773307f43d304f5b2e (patch)
tree57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libSBRdec
downloadODR-AudioEnc-2228e360595641dd906bf1773307f43d304f5b2e.tar.gz
ODR-AudioEnc-2228e360595641dd906bf1773307f43d304f5b2e.tar.bz2
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Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db
Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
Diffstat (limited to 'libSBRdec')
-rw-r--r--libSBRdec/include/sbrdecoder.h337
-rw-r--r--libSBRdec/src/arm/env_calc_arm.cpp148
-rw-r--r--libSBRdec/src/arm/lpp_tran_arm.cpp151
-rw-r--r--libSBRdec/src/env_calc.cpp2229
-rw-r--r--libSBRdec/src/env_calc.h165
-rw-r--r--libSBRdec/src/env_dec.cpp852
-rw-r--r--libSBRdec/src/env_dec.h101
-rw-r--r--libSBRdec/src/env_extr.cpp1395
-rw-r--r--libSBRdec/src/env_extr.h319
-rw-r--r--libSBRdec/src/huff_dec.cpp120
-rw-r--r--libSBRdec/src/huff_dec.h100
-rw-r--r--libSBRdec/src/lpp_tran.cpp1000
-rw-r--r--libSBRdec/src/lpp_tran.h242
-rw-r--r--libSBRdec/src/psbitdec.cpp593
-rw-r--r--libSBRdec/src/psbitdec.h103
-rw-r--r--libSBRdec/src/psdec.cpp1414
-rw-r--r--libSBRdec/src/psdec.h352
-rw-r--r--libSBRdec/src/psdec_hybrid.cpp652
-rw-r--r--libSBRdec/src/psdec_hybrid.h165
-rw-r--r--libSBRdec/src/sbr_crc.cpp183
-rw-r--r--libSBRdec/src/sbr_crc.h123
-rw-r--r--libSBRdec/src/sbr_deb.cpp90
-rw-r--r--libSBRdec/src/sbr_deb.h94
-rw-r--r--libSBRdec/src/sbr_dec.cpp1046
-rw-r--r--libSBRdec/src/sbr_dec.h210
-rw-r--r--libSBRdec/src/sbr_ram.cpp194
-rw-r--r--libSBRdec/src/sbr_ram.h158
-rw-r--r--libSBRdec/src/sbr_rom.cpp1412
-rw-r--r--libSBRdec/src/sbr_rom.h232
-rw-r--r--libSBRdec/src/sbr_scale.h123
-rw-r--r--libSBRdec/src/sbrdec_drc.cpp512
-rw-r--r--libSBRdec/src/sbrdec_drc.h150
-rw-r--r--libSBRdec/src/sbrdec_freq_sca.cpp805
-rw-r--r--libSBRdec/src/sbrdec_freq_sca.h107
-rw-r--r--libSBRdec/src/sbrdecoder.cpp1527
-rw-r--r--libSBRdec/src/transcendent.h355
36 files changed, 17759 insertions, 0 deletions
diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h
new file mode 100644
index 0000000..300a6d9
--- /dev/null
+++ b/libSBRdec/include/sbrdecoder.h
@@ -0,0 +1,337 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/************************ Fraunhofer IIS SBR decoder library ******************
+
+ Author(s):
+ Description: SBR decoder front-end prototypes and definitions.
+
+******************************************************************************/
+
+#ifndef __SBRDECODER_H
+#define __SBRDECODER_H
+
+#include "common_fix.h"
+
+#include "FDK_bitstream.h"
+#include "FDK_audio.h"
+
+
+#define SBR_DEBUG_EXTHLP "\
+--- SBR ---\n\
+ 0x00000010 Ancillary data and SBR-Header\n\
+ 0x00000020 SBR-Side info\n\
+ 0x00000040 Decoded SBR-bitstream data, e.g. envelope data\n\
+ 0x00000080 SBR-Bitstream statistics\n\
+ 0x00000100 Miscellaneous SBR-messages\n\
+ 0x00000200 SBR-Energies and gains in the adjustor\n\
+ 0x00000400 Fatal SBR errors\n\
+ 0x00000800 Transposer coefficients for inverse filtering\n\
+"
+
+/* Capability flags */
+#define CAPF_SBR_LP 0x00000001 /*!< Flag indicating library's capability of Low Power mode. */
+#define CAPF_SBR_HQ 0x00000002 /*!< Flag indicating library's capability of High Quality mode. */
+#define CAPF_SBR_DRM_BS 0x00000004 /*!< Flag indicating library's capability to decode DRM SBR data. */
+#define CAPF_SBR_CONCEALMENT 0x00000008 /*!< Flag indicating library's capability to conceal erroneous frames. */
+#define CAPF_SBR_DRC 0x00000010 /*!< Flag indicating library's capability for Dynamic Range Control. */
+#define CAPF_SBR_PS_MPEG 0x00000020 /*!< Flag indicating library's capability to do MPEG Parametric Stereo. */
+#define CAPF_SBR_PS_DRM 0x00000040 /*!< Flag indicating library's capability to do DRM Parametric Stereo. */
+
+typedef enum
+{
+ SBRDEC_OK = 0, /*!< All fine. */
+ /* SBRDEC_CONCEAL, */
+ /* SBRDEC_NOSYNCH, */
+ /* SBRDEC_ILLEGAL_PROGRAM, */
+ /* SBRDEC_ILLEGAL_TAG, */
+ /* SBRDEC_ILLEGAL_CHN_CONFIG, */
+ /* SBRDEC_ILLEGAL_SECTION, */
+ /* SBRDEC_ILLEGAL_SCFACTORS, */
+ /* SBRDEC_ILLEGAL_PULSE_DATA, */
+ /* SBRDEC_MAIN_PROFILE_NOT_IMPLEMENTED, */
+ /* SBRDEC_GC_NOT_IMPLEMENTED, */
+ /* SBRDEC_ILLEGAL_PLUS_ELE_ID, */
+ SBRDEC_CREATE_ERROR, /*!< */
+ SBRDEC_NOT_INITIALIZED, /*!< */
+ SBRDEC_MEM_ALLOC_FAILED, /*!< Memory allocation failed. Probably not enough memory available. */
+ SBRDEC_PARSE_ERROR, /*!< */
+ SBRDEC_UNSUPPORTED_CONFIG, /*!< */
+ SBRDEC_SET_PARAM_FAIL /*!< */
+} SBR_ERROR;
+
+typedef enum
+{
+ SBR_SYSTEM_BITSTREAM_DELAY, /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */
+ SBR_QMF_MODE, /*!< Set QMF mode, either complex or low power. */
+ SBR_LD_QMF_TIME_ALIGN, /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */
+ SBR_BS_INTERRUPTION /*!< Signal bit stream interruption. Value is ignored. */
+} SBRDEC_PARAM;
+
+typedef struct SBR_DECODER_INSTANCE *HANDLE_SBRDECODER;
+
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+
+/**
+ * \brief Allocates and initializes one SBR decoder instance.
+ * \param pSelf Pointer to where a SBR decoder handle is copied into.
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER *pSelf );
+
+/**
+ * \brief Initialize a SBR decoder runtime instance. Must be called before decoding starts.
+ *
+ * \param self Handle to a SBR decoder instance.
+ * \param sampleRateIn Input samplerate of the SBR decoder instance.
+ * \param sampleRateOut Output samplerate of the SBR decoder instance.
+ * \param samplesPerFrame Number of samples per frames.
+ * \param coreCodec Audio Object Type (AOT) of the core codec.
+ * \param elementID Table with MPEG-4 element Ids in canonical order.
+ * \param forceReset Flag that enforces a complete decoder reset.
+ *
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_InitElement (
+ HANDLE_SBRDECODER self,
+ const int sampleRateIn,
+ const int sampleRateOut,
+ const int samplesPerFrame,
+ const AUDIO_OBJECT_TYPE coreCodec,
+ const MP4_ELEMENT_ID elementID,
+ const int elementIndex
+ );
+
+/**
+ * \brief pass out of band SBR header to SBR decoder
+ *
+ * \param self Handle to a SBR decoder instance.
+ * \param hBs bit stream handle data source.
+ * \param elementID SBR element ID.
+ * \param elementIndex SBR element index.
+ *
+ * \return Error code.
+ */
+INT sbrDecoder_Header (
+ HANDLE_SBRDECODER self,
+ HANDLE_FDK_BITSTREAM hBs,
+ const INT sampleRateIn,
+ const INT sampleRateOut,
+ const INT samplesPerFrame,
+ const AUDIO_OBJECT_TYPE coreCodec,
+ const MP4_ELEMENT_ID elementID,
+ const INT elementIndex
+ );
+
+/**
+ * \brief Set a parameter of the SBR decoder runtime instance.
+ * \param self SBR decoder handle.
+ * \param param Parameter which will be set if successfull.
+ * \param value New parameter value.
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_SetParam ( HANDLE_SBRDECODER self,
+ const SBRDEC_PARAM param,
+ const INT value );
+
+/**
+ * \brief Feed DRC channel data into a SBR decoder runtime instance.
+ *
+ * \param self SBR decoder handle.
+ * \param ch Channel number to which the DRC data is associated to.
+ * \param numBands Number of DRC bands.
+ * \param pNextFact_mag Pointer to a table with the DRC factor magnitudes.
+ * \param nextFact_exp Exponent for all DRC factors.
+ * \param drcInterpolationScheme DRC interpolation scheme.
+ * \param winSequence Window sequence from core coder (eight short or one long window).
+ * \param pBandTop Pointer to a table with the top borders for all DRC bands.
+ *
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER self,
+ INT ch,
+ UINT numBands,
+ FIXP_DBL *pNextFact_mag,
+ INT nextFact_exp,
+ SHORT drcInterpolationScheme,
+ UCHAR winSequence,
+ USHORT *pBandTop );
+
+/**
+ * \brief Disable SBR DRC for a certain channel.
+ *
+ * \param hSbrDecoder SBR decoder handle.
+ * \param ch Number of the channel that has to be disabled.
+ *
+ * \return None.
+ */
+void sbrDecoder_drcDisable ( HANDLE_SBRDECODER self,
+ INT ch );
+
+
+/**
+ * \brief Parse one SBR element data extension data block. The bit stream position will
+ * be placed at the end of the SBR payload block. The remaining bits will be returned
+ * into *count if a payload length is given (byPayLen > 0). If no SBR payload length is
+ * given (bsPayLen < 0) then the bit stream position on return will be random after this
+ * function call in case of errors, and any further decoding will be completely pointless.
+ *
+ * \param self SBR decoder handle.
+ * \param hBs Bit stream handle as data source.
+ * \param count Pointer to an integer where the amount of parsed SBR payload bits is stored into.
+ * \param bsPayLen If > 0 this value is the SBR payload length. If < 0, the SBR payload length is unknown.
+ * \param flags CRC flag (0: EXT_SBR_DATA; 1: EXT_SBR_DATA_CRC)
+ * \param prev_element Previous MPEG-4 element ID.
+ * \param element_index Index of the current element.
+ *
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_Parse (
+ HANDLE_SBRDECODER self,
+ HANDLE_FDK_BITSTREAM hBs,
+ int *count,
+ int bsPayLen,
+ int crcFlag,
+ MP4_ELEMENT_ID prev_element,
+ int element_index,
+ int fGlobalIndependencyFlag
+ );
+
+/**
+ * \brief This function decodes the given SBR bitstreams and applies SBR to the given time data.
+ *
+ * SBR-processing works InPlace. I.e. the calling function has to provide
+ * a time domain buffer timeData which can hold the completely decoded
+ * result.
+ *
+ * Left and right channel are read and stored according to the
+ * interleaving flag, frame length and number of channels.
+ *
+ * \param self Handle of an open SBR decoder instance.
+ * \param hSbrBs SBR Bitstream handle.
+ * \param timeData Pointer to input and finally upsampled output data.
+ * \param numChannels Pointer to a buffer holding the number of channels in time data buffer.
+ * \param sampleRate Output samplerate.
+ * \param channelMapping Channel mapping indices.
+ * \param interleaved Flag indicating if time data is stored interleaved (1: Interleaved time data, 0: non-interleaved timedata).
+ * \param coreDecodedOk Flag indicating if the core decoder did not find any error (0: core decoder found errors, 1: no errors).
+ * \param psDecoded Pointer to a buffer holding a flag. Input: PS is possible, Output: PS has been rendered.
+ *
+ * \return Error code.
+ */
+SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER self,
+ INT_PCM *timeData,
+ int *numChannels,
+ int *sampleRate,
+ const UCHAR channelMapping[(6)],
+ const int interleaved,
+ const int coreDecodedOk,
+ UCHAR *psDecoded );
+
+
+/**
+ * \brief Close SBR decoder instance and free memory.
+ * \param self SBR decoder handle.
+ * \return Error Code.
+ */
+SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *self );
+
+
+/**
+ * \brief Get SBR decoder library information.
+ * \param info Pointer to a LIB_INFO struct, where library information is written to.
+ * \return 0 on success, -1 if invalid handle or if no free element is available to write information to.
+ */
+INT sbrDecoder_GetLibInfo( LIB_INFO *info );
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/libSBRdec/src/arm/env_calc_arm.cpp b/libSBRdec/src/arm/env_calc_arm.cpp
new file mode 100644
index 0000000..4e3a6de
--- /dev/null
+++ b/libSBRdec/src/arm/env_calc_arm.cpp
@@ -0,0 +1,148 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** Fraunhofer IIS ***************************
+
+ Author(s): Arthur Tritthart
+ Description: (ARM optimised) SBR domain coding
+
+******************************************************************************/
+#ifndef INCLUSION_GUARD_CALC_ENV_ARM
+#define INCLUSION_GUARD_CALC_ENV_ARM
+
+
+/*!
+ \brief Compute maximal value of a complex array (re/im) of a given width
+ Negative values are temporarily logically or'ed with 0xFFFFFFFF
+ instead of negating the value, if the sign bit is set.
+ \param maxVal Preset maximal value
+ \param reTmp real input signal
+ \param imTmp imaginary input signal
+ \return new maximal value
+*/
+
+#ifdef FUNCTION_FDK_get_maxval
+__asm FIXP_DBL FDK_get_maxval (FIXP_DBL maxVal, FIXP_DBL *reTmp, FIXP_DBL *imTmp, int width )
+{
+
+ /* Register map:
+ r0 maxVal
+ r1 reTmp
+ r2 imTmp
+ r3 width
+ r4 real
+ r5 imag
+ */
+ PUSH {r4-r5}
+
+ MOVS r3, r3, ASR #1
+ ADC r3, r3, #0
+ BCS FDK_get_maxval_loop_2nd_part
+ BEQ FDK_get_maxval_loop_end
+
+FDK_get_maxval_loop
+ LDR r4, [r1], #4
+ LDR r5, [r2], #4
+ EOR r4, r4, r4, ASR #31
+ EOR r5, r5, r5, ASR #31
+ ORR r0, r0, r4
+ ORR r0, r0, r5
+
+FDK_get_maxval_loop_2nd_part
+ LDR r4, [r1], #4
+ LDR r5, [r2], #4
+ EOR r4, r4, r4, ASR #31
+ EOR r5, r5, r5, ASR #31
+ ORR r0, r0, r4
+ ORR r0, r0, r5
+
+ SUBS r3, r3, #1
+ BNE FDK_get_maxval_loop
+
+FDK_get_maxval_loop_end
+ POP {r4-r5}
+ BX lr
+}
+#endif /* FUNCTION_FDK_get_maxval */
+
+#endif /* INCLUSION_GUARD_CALC_ENV_ARM */
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
new file mode 100644
index 0000000..541e7c7
--- /dev/null
+++ b/libSBRdec/src/arm/lpp_tran_arm.cpp
@@ -0,0 +1,151 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/******************************** Fraunhofer IIS ***************************
+
+ Author(s): Arthur Tritthart
+ Description: (ARM optimised) LPP transposer subroutines
+
+******************************************************************************/
+
+
+#if defined(__arm__)
+
+
+#define FUNCTION_LPPTRANSPOSER_func1
+
+#ifdef FUNCTION_LPPTRANSPOSER_func1
+
+/* Note: This code requires only 43 cycles per iteration instead of 61 on ARM926EJ-S */
+__attribute__ ((noinline)) static void lppTransposer_func1(
+ FIXP_DBL *lowBandReal,
+ FIXP_DBL *lowBandImag,
+ FIXP_DBL **qmfBufferReal,
+ FIXP_DBL **qmfBufferImag,
+ int loops,
+ int hiBand,
+ int dynamicScale,
+ int descale,
+ FIXP_SGL a0r,
+ FIXP_SGL a0i,
+ FIXP_SGL a1r,
+ FIXP_SGL a1i)
+{
+
+ FIXP_DBL real1, real2, imag1, imag2, accu1, accu2;
+
+ real2 = lowBandReal[-2];
+ real1 = lowBandReal[-1];
+ imag2 = lowBandImag[-2];
+ imag1 = lowBandImag[-1];
+ for(int i=0; i < loops; i++)
+ {
+ accu1 = fMultDiv2( a0r,real1);
+ accu2 = fMultDiv2( a0i,imag1);
+ accu1 = fMultAddDiv2(accu1,a1r,real2);
+ accu2 = fMultAddDiv2(accu2,a1i,imag2);
+ real2 = fMultDiv2( a1i,real2);
+ accu1 = accu1 - accu2;
+ accu1 = accu1 >> dynamicScale;
+
+ accu2 = fMultAddDiv2(real2,a1r,imag2);
+ real2 = real1;
+ imag2 = imag1;
+ accu2 = fMultAddDiv2(accu2,a0i,real1);
+ real1 = lowBandReal[i];
+ accu2 = fMultAddDiv2(accu2,a0r,imag1);
+ imag1 = lowBandImag[i];
+ accu2 = accu2 >> dynamicScale;
+
+ accu1 <<= 1;
+ accu2 <<= 1;
+
+ qmfBufferReal[i][hiBand] = accu1 + (real1>>descale);
+ qmfBufferImag[i][hiBand] = accu2 + (imag1>>descale);
+ }
+}
+#endif /* #ifdef FUNCTION_LPPTRANSPOSER_func1 */
+#endif /* __arm__ */
+
+
+
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
new file mode 100644
index 0000000..11df761
--- /dev/null
+++ b/libSBRdec/src/env_calc.cpp
@@ -0,0 +1,2229 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Envelope calculation
+
+ The envelope adjustor compares the energies present in the transposed
+ highband to the reference energies conveyed with the bitstream.
+ The highband is amplified (sometimes) or attenuated (mostly) to the
+ desired level.
+
+ The spectral shape of the reference energies can be changed several times per
+ frame if necessary. Each set of energy values corresponding to a certain range
+ in time will be called an <em>envelope</em> here.
+ The bitstream supports several frequency scales and two resolutions. Normally,
+ one or more QMF-subbands are grouped to one SBR-band. An envelope contains
+ reference energies for each SBR-band.
+ In addition to the energy envelopes, noise envelopes are transmitted that
+ define the ratio of energy which is generated by adding noise instead of
+ transposing the lowband. The noise envelopes are given in a coarser time
+ and frequency resolution.
+ If a signal contains strong tonal components, synthetic sines can be
+ generated in individual SBR bands.
+
+ An overlap buffer of 6 QMF-timeslots is used to allow a more
+ flexible alignment of the envelopes in time that is not restricted to the
+ core codec's frame borders.
+ Therefore the envelope adjustor has access to the spectral data of the
+ current frame as well as the last 6 QMF-timeslots of the previous frame.
+ However, in average only the data of 1 frame is being processed as
+ the adjustor is called once per frame.
+
+ Depending on the frequency range set in the bitstream, only QMF-subbands between
+ <em>lowSubband</em> and <em>highSubband</em> are adjusted.
+
+ Scaling of spectral data to maximize SNR (see #QMF_SCALE_FACTOR) as well as a special Mantissa-Exponent format
+ ( see calculateSbrEnvelope() ) are being used. The main entry point for this modules is calculateSbrEnvelope().
+
+ \sa sbr_scale.h, #QMF_SCALE_FACTOR, calculateSbrEnvelope(), \ref documentationOverview
+*/
+
+
+#include "env_calc.h"
+
+#include "sbrdec_freq_sca.h"
+#include "env_extr.h"
+#include "transcendent.h"
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h" /* need FDKpow() for debug outputs */
+
+#if defined(__arm__)
+#include "arm/env_calc_arm.cpp"
+#endif
+
+typedef struct
+{
+ FIXP_DBL nrgRef[MAX_FREQ_COEFFS];
+ FIXP_DBL nrgEst[MAX_FREQ_COEFFS];
+ FIXP_DBL nrgGain[MAX_FREQ_COEFFS];
+ FIXP_DBL noiseLevel[MAX_FREQ_COEFFS];
+ FIXP_DBL nrgSine[MAX_FREQ_COEFFS];
+
+ SCHAR nrgRef_e[MAX_FREQ_COEFFS];
+ SCHAR nrgEst_e[MAX_FREQ_COEFFS];
+ SCHAR nrgGain_e[MAX_FREQ_COEFFS];
+ SCHAR noiseLevel_e[MAX_FREQ_COEFFS];
+ SCHAR nrgSine_e[MAX_FREQ_COEFFS];
+}
+ENV_CALC_NRGS;
+
+/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
+ SCHAR *filtBuffer_e,
+ FIXP_DBL *NrgGain,
+ SCHAR *NrgGain_e,
+ int subbands);
+
+/*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal,
+ FIXP_DBL **analysBufferImag,
+ int lowSubband, int highSubband,
+ int start_pos, int next_pos,
+ SCHAR frameExp,
+ FIXP_DBL *nrgEst,
+ SCHAR *nrgEst_e );
+
+/*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal,
+ FIXP_DBL **analysBufferImag,
+ int nSfb,
+ UCHAR *freqBandTable,
+ int start_pos, int next_pos,
+ SCHAR input_e,
+ FIXP_DBL *nrg_est,
+ SCHAR *nrg_est_e );
+
+/*static*/ void calcSubbandGain(FIXP_DBL nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
+ FIXP_DBL tmpNoise, SCHAR tmpNoise_e,
+ UCHAR sinePresentFlag,
+ UCHAR sineMapped,
+ int noNoiseFlag);
+
+/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+ int lowSubband,
+ int highSubband,
+ FIXP_DBL *sumRef_m,
+ SCHAR *sumRef_e,
+ FIXP_DBL *ptrAvgGain_m,
+ SCHAR *ptrAvgGain_e);
+
+/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,
+ ENV_CALC_NRGS* nrgs,
+ UCHAR *ptrHarmIndex,
+ int lowSubbands,
+ int noSubbands,
+ int scale_change,
+ int noNoiseFlag,
+ int *ptrPhaseIndex,
+ int fCldfb);
+/*static*/ void adjustTimeSlotHQ(FIXP_DBL *ptrReal,
+ FIXP_DBL *ptrImag,
+ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+ ENV_CALC_NRGS* nrgs,
+ int lowSubbands,
+ int noSubbands,
+ int scale_change,
+ FIXP_SGL smooth_ratio,
+ int noNoiseFlag,
+ int filtBufferNoiseShift);
+
+
+/*!
+ \brief Map sine flags from bitstream to QMF bands
+
+ The bitstream carries only 1 sine flag per band and frame.
+ This function maps every sine flag from the bitstream to a specific QMF subband
+ and to a specific envelope where the sine shall start.
+ The result is stored in the vector sineMapped which contains one entry per
+ QMF subband. The value of an entry specifies the envelope where a sine
+ shall start. A value of #MAX_ENVELOPES indicates that no sine is present
+ in the subband.
+ The missing harmonics flags from the previous frame (harmFlagsPrev) determine
+ if a sine starts at the beginning of the frame or at the transient position.
+ Additionally, the flags in harmFlagsPrev are being updated by this function
+ for the next frame.
+*/
+/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
+ int nSfb, /*!< Number of bands in the table */
+ UCHAR *addHarmonics, /*!< vector with 1 flag per sfb */
+ int *harmFlagsPrev, /*!< Packed 'addHarmonics' */
+ int tranEnv, /*!< Transient position */
+ SCHAR *sineMapped) /*!< Resulting vector of sine start positions for each QMF band */
+
+{
+ int i;
+ int lowSubband2 = freqBandTable[0]<<1;
+ int bitcount = 0;
+ int oldflags = *harmFlagsPrev;
+ int newflags = 0;
+
+ /*
+ Format of harmFlagsPrev:
+
+ first word = flags for highest 16 sfb bands in use
+ second word = flags for next lower 16 sfb bands (if present)
+ third word = flags for lowest 16 sfb bands (if present)
+
+ Up to MAX_FREQ_COEFFS sfb bands can be flagged for a sign.
+ The lowest bit of the first word corresponds to the _highest_ sfb band in use.
+ This is ensures that each flag is mapped to the same QMF band even after a
+ change of the crossover-frequency.
+ */
+
+
+ /* Reset the output vector first */
+ FDKmemset(sineMapped, MAX_ENVELOPES,MAX_FREQ_COEFFS); /* MAX_ENVELOPES means 'no sine' */
+
+ freqBandTable += nSfb;
+ addHarmonics += nSfb-1;
+
+ for (i=nSfb; i!=0; i--) {
+ int ui = *freqBandTable--; /* Upper limit of the current scale factor band. */
+ int li = *freqBandTable; /* Lower limit of the current scale factor band. */
+
+ if ( *addHarmonics-- ) { /* There is a sine in this band */
+
+ unsigned int mask = 1 << bitcount;
+ newflags |= mask; /* Set flag */
+
+ /*
+ If there was a sine in the last frame, let it continue from the first envelope on
+ else start at the transient position.
+ */
+ sineMapped[(ui+li-lowSubband2) >> 1] = ( oldflags & mask ) ? 0 : tranEnv;
+ }
+
+ if ((++bitcount == 16) || i==1) {
+ bitcount = 0;
+ *harmFlagsPrev++ = newflags;
+ oldflags = *harmFlagsPrev; /* Fetch 16 of the old flags */
+ newflags = 0;
+ }
+ }
+}
+
+
+/*!
+ \brief Reduce gain-adjustment induced aliasing for real valued filterbank.
+*/
+/*static*/ void
+aliasingReduction(FIXP_DBL* degreeAlias, /*!< estimated aliasing for each QMF channel */
+ ENV_CALC_NRGS* nrgs,
+ int* useAliasReduction, /*!< synthetic sine engergy for each subband, used as flag */
+ int noSubbands) /*!< number of QMF channels to process */
+{
+ FIXP_DBL* nrgGain = nrgs->nrgGain; /*!< subband gains to be modified */
+ SCHAR* nrgGain_e = nrgs->nrgGain_e; /*!< subband gains to be modified (exponents) */
+ FIXP_DBL* nrgEst = nrgs->nrgEst; /*!< subband energy before amplification */
+ SCHAR* nrgEst_e = nrgs->nrgEst_e; /*!< subband energy before amplification (exponents) */
+ int grouping = 0, index = 0, noGroups, k;
+ int groupVector[MAX_FREQ_COEFFS];
+
+ /* Calculate grouping*/
+ for (k = 0; k < noSubbands-1; k++ ){
+ if ( (degreeAlias[k + 1] != FL2FXCONST_DBL(0.0f)) && useAliasReduction[k] ) {
+ if(grouping==0){
+ groupVector[index++] = k;
+ grouping = 1;
+ }
+ else{
+ if(groupVector[index-1] + 3 == k){
+ groupVector[index++] = k + 1;
+ grouping = 0;
+ }
+ }
+ }
+ else{
+ if(grouping){
+ if(useAliasReduction[k])
+ groupVector[index++] = k + 1;
+ else
+ groupVector[index++] = k;
+ grouping = 0;
+ }
+ }
+ }
+
+ if(grouping){
+ groupVector[index++] = noSubbands;
+ }
+ noGroups = index >> 1;
+
+
+ /*Calculate new gain*/
+ for (int group = 0; group < noGroups; group ++) {
+ FIXP_DBL nrgOrig = FL2FXCONST_DBL(0.0f); /* Original signal energy in current group of bands */
+ SCHAR nrgOrig_e = 0;
+ FIXP_DBL nrgAmp = FL2FXCONST_DBL(0.0f); /* Amplified signal energy in group (using current gains) */
+ SCHAR nrgAmp_e = 0;
+ FIXP_DBL nrgMod = FL2FXCONST_DBL(0.0f); /* Signal energy in group when applying modified gains */
+ SCHAR nrgMod_e = 0;
+ FIXP_DBL groupGain; /* Total energy gain in group */
+ SCHAR groupGain_e;
+ FIXP_DBL compensation; /* Compensation factor for the energy change when applying modified gains */
+ SCHAR compensation_e;
+
+ int startGroup = groupVector[2*group];
+ int stopGroup = groupVector[2*group+1];
+
+ /* Calculate total energy in group before and after amplification with current gains: */
+ for(k = startGroup; k < stopGroup; k++){
+ /* Get original band energy */
+ FIXP_DBL tmp = nrgEst[k];
+ SCHAR tmp_e = nrgEst_e[k];
+
+ FDK_add_MantExp(tmp, tmp_e, nrgOrig, nrgOrig_e, &nrgOrig, &nrgOrig_e);
+
+ /* Multiply band energy with current gain */
+ tmp = fMult(tmp,nrgGain[k]);
+ tmp_e = tmp_e + nrgGain_e[k];
+
+ FDK_add_MantExp(tmp, tmp_e, nrgAmp, nrgAmp_e, &nrgAmp, &nrgAmp_e);
+ }
+
+ /* Calculate total energy gain in group */
+ FDK_divide_MantExp(nrgAmp, nrgAmp_e,
+ nrgOrig, nrgOrig_e,
+ &groupGain, &groupGain_e);
+
+ for(k = startGroup; k < stopGroup; k++){
+ FIXP_DBL tmp;
+ SCHAR tmp_e;
+
+ FIXP_DBL alpha = degreeAlias[k];
+ if (k < noSubbands - 1) {
+ if (degreeAlias[k + 1] > alpha)
+ alpha = degreeAlias[k + 1];
+ }
+
+ /* Modify gain depending on the degree of aliasing */
+ FDK_add_MantExp( fMult(alpha,groupGain), groupGain_e,
+ fMult(/*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - alpha,nrgGain[k]), nrgGain_e[k],
+ &nrgGain[k], &nrgGain_e[k] );
+
+ /* Apply modified gain to original energy */
+ tmp = fMult(nrgGain[k],nrgEst[k]);
+ tmp_e = nrgGain_e[k] + nrgEst_e[k];
+
+ /* Accumulate energy with modified gains applied */
+ FDK_add_MantExp( tmp, tmp_e,
+ nrgMod, nrgMod_e,
+ &nrgMod, &nrgMod_e );
+ }
+
+ /* Calculate compensation factor to retain the energy of the amplified signal */
+ FDK_divide_MantExp(nrgAmp, nrgAmp_e,
+ nrgMod, nrgMod_e,
+ &compensation, &compensation_e);
+
+ /* Apply compensation factor to all gains of the group */
+ for(k = startGroup; k < stopGroup; k++){
+ nrgGain[k] = fMult(nrgGain[k],compensation);
+ nrgGain_e[k] = nrgGain_e[k] + compensation_e;
+ }
+ }
+}
+
+
+ /* Convert headroom bits to exponent */
+#define SCALE2EXP(s) (15-(s))
+#define EXP2SCALE(e) (15-(e))
+
+/*!
+ \brief Apply spectral envelope to subband samples
+
+ This function is called from sbr_dec.cpp in each frame.
+
+ To enhance accuracy and due to the usage of tables for squareroots and
+ inverse, some calculations are performed with the operands being split
+ into mantissa and exponent. The variable names in the source code carry
+ the suffixes <em>_m</em> and <em>_e</em> respectively. The control data
+ in #hFrameData containts envelope data which is represented by this format but
+ stored in single words. (See requantizeEnvelopeData() for details). This data
+ is unpacked within calculateSbrEnvelope() to follow the described suffix convention.
+
+ The actual value (comparable to the corresponding float-variable in the
+ research-implementation) of a mantissa/exponent-pair can be calculated as
+
+ \f$ value = value\_m * 2^{value\_e} \f$
+
+ All energies and noise levels decoded from the bitstream suit for an
+ original signal magnitude of \f$\pm 32768 \f$ rather than \f$ \pm 1\f$. Therefore,
+ the scale factor <em>hb_scale</em> passed into this function will be converted
+ to an 'input exponent' (#input_e), which fits the internal representation.
+
+ Before the actual processing, an exponent #adj_e for resulting adjusted
+ samples is derived from the maximum reference energy.
+
+ Then, for each envelope, the following steps are performed:
+
+ \li Calculate energy in the signal to be adjusted. Depending on the the value of
+ #interpolFreq (interpolation mode), this is either done seperately
+ for each QMF-subband or for each SBR-band.
+ The resulting energies are stored in #nrgEst_m[#MAX_FREQ_COEFFS] (mantissas)
+ and #nrgEst_e[#MAX_FREQ_COEFFS] (exponents).
+ \li Calculate gain and noise level for each subband:<br>
+ \f$ gain = \sqrt{ \frac{nrgRef}{nrgEst} \cdot (1 - noiseRatio) }
+ \hspace{2cm}
+ noise = \sqrt{ nrgRef \cdot noiseRatio }
+ \f$<br>
+ where <em>noiseRatio</em> and <em>nrgRef</em> are extracted from the
+ bitstream and <em>nrgEst</em> is the subband energy before adjustment.
+ The resulting gains are stored in #nrgGain_m[#MAX_FREQ_COEFFS]
+ (mantissas) and #nrgGain_e[#MAX_FREQ_COEFFS] (exponents), the noise levels
+ are stored in #noiseLevel_m[#MAX_FREQ_COEFFS] and #noiseLevel_e[#MAX_FREQ_COEFFS]
+ (exponents).
+ The sine levels are stored in #nrgSine_m[#MAX_FREQ_COEFFS]
+ and #nrgSine_e[#MAX_FREQ_COEFFS].
+ \li Noise limiting: The gain for each subband is limited both absolutely
+ and relatively compared to the total gain over all subbands.
+ \li Boost gain: Calculate and apply boost factor for each limiter band
+ in order to compensate for the energy loss imposed by the limiting.
+ \li Apply gains and add noise: The gains and noise levels are applied
+ to all timeslots of the current envelope. A short FIR-filter (length 4
+ QMF-timeslots) can be used to smooth the sudden change at the envelope borders.
+ Each complex subband sample of the current timeslot is multiplied by the
+ smoothed gain, then random noise with the calculated level is added.
+
+ \note
+ To reduce the stack size, some of the local arrays could be located within
+ the time output buffer. Of the 512 samples temporarily available there,
+ about half the size is already used by #SBR_FRAME_DATA. A pointer to the
+ remaining free memory could be supplied by an additional argument to calculateSbrEnvelope()
+ in sbr_dec:
+
+ \par
+ \code
+ calculateSbrEnvelope (&hSbrDec->sbrScaleFactor,
+ &hSbrDec->SbrCalculateEnvelope,
+ hHeaderData,
+ hFrameData,
+ QmfBufferReal,
+ QmfBufferImag,
+ timeOutPtr + sizeof(SBR_FRAME_DATA)/sizeof(Float) + 1);
+ \endcode
+
+ \par
+ Within calculateSbrEnvelope(), some pointers could be defined instead of the arrays
+ #nrgRef_m, #nrgRef_e, #nrgEst_m, #nrgEst_e, #noiseLevel_m:
+
+ \par
+ \code
+ fract* nrgRef_m = timeOutPtr;
+ SCHAR* nrgRef_e = nrgRef_m + MAX_FREQ_COEFFS;
+ fract* nrgEst_m = nrgRef_e + MAX_FREQ_COEFFS;
+ SCHAR* nrgEst_e = nrgEst_m + MAX_FREQ_COEFFS;
+ fract* noiseLevel_m = nrgEst_e + MAX_FREQ_COEFFS;
+ \endcode
+
+ <br>
+*/
+void
+calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */
+ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env, /*!< Handle to struct filled by the create-function */
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */
+ FIXP_DBL **analysBufferReal, /*!< Real part of subband samples to be processed */
+ FIXP_DBL **analysBufferImag, /*!< Imag part of subband samples to be processed */
+ const int useLP,
+ FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */
+ const UINT flags,
+ const int frameErrorFlag
+ )
+{
+ int c, i, j, envNoise = 0;
+ UCHAR* borders = hFrameData->frameInfo.borders;
+
+ FIXP_SGL *noiseLevels = hFrameData->sbrNoiseFloorLevel;
+ HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+
+ int lowSubband = hFreq->lowSubband;
+ int highSubband = hFreq->highSubband;
+ int noSubbands = highSubband - lowSubband;
+
+ int noNoiseBands = hFreq->nNfb;
+ int no_cols = hHeaderData->numberTimeSlots * hHeaderData->timeStep;
+ UCHAR first_start = borders[0] * hHeaderData->timeStep;
+
+ SCHAR sineMapped[MAX_FREQ_COEFFS];
+ SCHAR ov_adj_e = SCALE2EXP(sbrScaleFactor->ov_hb_scale);
+ SCHAR adj_e = 0;
+ SCHAR output_e;
+ SCHAR final_e = 0;
+
+ SCHAR maxGainLimit_e = (frameErrorFlag) ? MAX_GAIN_CONCEAL_EXP : MAX_GAIN_EXP;
+
+ int useAliasReduction[64];
+ UCHAR smooth_length = 0;
+
+ FIXP_SGL * pIenv = hFrameData->iEnvelope;
+
+ /*
+ Extract sine flags for all QMF bands
+ */
+ mapSineFlags(hFreq->freqBandTable[1],
+ hFreq->nSfb[1],
+ hFrameData->addHarmonics,
+ h_sbr_cal_env->harmFlagsPrev,
+ hFrameData->frameInfo.tranEnv,
+ sineMapped);
+
+
+ /*
+ Scan for maximum in bufferd noise levels.
+ This is needed in case that we had strong noise in the previous frame
+ which is smoothed into the current frame.
+ The resulting exponent is used as start value for the maximum search
+ in reference energies
+ */
+ if (!useLP)
+ adj_e = h_sbr_cal_env->filtBufferNoise_e - getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+
+ /*
+ Scan for maximum reference energy to be able
+ to select appropriate values for adj_e and final_e.
+ */
+
+ for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {
+ INT maxSfbNrg_e = -FRACT_BITS+NRG_EXP_OFFSET; /* start value for maximum search */
+
+ /* Fetch frequency resolution for current envelope: */
+ for (j=hFreq->nSfb[hFrameData->frameInfo.freqRes[i]]; j!=0; j--) {
+ maxSfbNrg_e = fixMax(maxSfbNrg_e,(INT)((LONG)(*pIenv++) & MASK_E));
+ }
+ maxSfbNrg_e -= NRG_EXP_OFFSET;
+
+ /* Energy -> magnitude (sqrt halfens exponent) */
+ maxSfbNrg_e = (maxSfbNrg_e+1) >> 1; /* +1 to go safe (round to next higher int) */
+
+ /* Some safety margin is needed for 2 reasons:
+ - The signal energy is not equally spread over all subband samples in
+ a specific sfb of an envelope (Nrg could be too high by a factor of
+ envWidth * sfbWidth)
+ - Smoothing can smear high gains of the previous envelope into the current
+ */
+ maxSfbNrg_e += 6;
+
+ if (borders[i] < hHeaderData->numberTimeSlots)
+ /* This envelope affects timeslots that belong to the output frame */
+ adj_e = (maxSfbNrg_e > adj_e) ? maxSfbNrg_e : adj_e;
+
+ if (borders[i+1] > hHeaderData->numberTimeSlots)
+ /* This envelope affects timeslots after the output frame */
+ final_e = (maxSfbNrg_e > final_e) ? maxSfbNrg_e : final_e;
+
+ }
+
+ /*
+ Calculate adjustment factors and apply them for every envelope.
+ */
+ pIenv = hFrameData->iEnvelope;
+
+ for (i = 0; i < hFrameData->frameInfo.nEnvelopes; i++) {
+
+ int k, noNoiseFlag;
+ SCHAR noise_e, input_e = SCALE2EXP(sbrScaleFactor->hb_scale);
+ C_ALLOC_SCRATCH_START(pNrgs, ENV_CALC_NRGS, 1);
+
+ /*
+ Helper variables.
+ */
+ UCHAR start_pos = hHeaderData->timeStep * borders[i]; /* Start-position in time (subband sample) for current envelope. */
+ UCHAR stop_pos = hHeaderData->timeStep * borders[i+1]; /* Stop-position in time (subband sample) for current envelope. */
+ UCHAR freq_res = hFrameData->frameInfo.freqRes[i]; /* Frequency resolution for current envelope. */
+
+
+ /* Always do fully initialize the temporary energy table. This prevents negative energies and extreme gain factors in
+ cases where the number of limiter bands exceeds the number of subbands. The latter can be caused by undetected bit
+ errors and is tested by some streams from the certification set. */
+ FDKmemclear(pNrgs, sizeof(ENV_CALC_NRGS));
+
+ /* If the start-pos of the current envelope equals the stop pos of the current
+ noise envelope, increase the pointer (i.e. choose the next noise-floor).*/
+ if (borders[i] == hFrameData->frameInfo.bordersNoise[envNoise+1]){
+ noiseLevels += noNoiseBands; /* The noise floor data is stored in a row [noiseFloor1 noiseFloor2...].*/
+ envNoise++;
+ }
+
+ if(i==hFrameData->frameInfo.tranEnv || i==h_sbr_cal_env->prevTranEnv) /* attack */
+ {
+ noNoiseFlag = 1;
+ if (!useLP)
+ smooth_length = 0; /* No smoothing on attacks! */
+ }
+ else {
+ noNoiseFlag = 0;
+ if (!useLP)
+ smooth_length = (1 - hHeaderData->bs_data.smoothingLength) << 2; /* can become either 0 or 4 */
+ }
+
+
+ /*
+ Energy estimation in transposed highband.
+ */
+ if (hHeaderData->bs_data.interpolFreq)
+ calcNrgPerSubband(analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ lowSubband, highSubband,
+ start_pos, stop_pos,
+ input_e,
+ pNrgs->nrgEst,
+ pNrgs->nrgEst_e);
+ else
+ calcNrgPerSfb(analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ hFreq->nSfb[freq_res],
+ hFreq->freqBandTable[freq_res],
+ start_pos, stop_pos,
+ input_e,
+ pNrgs->nrgEst,
+ pNrgs->nrgEst_e);
+
+ /*
+ Calculate subband gains
+ */
+ {
+ UCHAR * table = hFreq->freqBandTable[freq_res];
+ UCHAR * pUiNoise = &hFreq->freqBandTableNoise[1]; /*! Upper limit of the current noise floor band. */
+
+ FIXP_SGL * pNoiseLevels = noiseLevels;
+
+ FIXP_DBL tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
+ SCHAR tmpNoise_e = (UCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;
+
+ int cc = 0;
+ c = 0;
+ for (j = 0; j < hFreq->nSfb[freq_res]; j++) {
+
+ FIXP_DBL refNrg = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pIenv) & MASK_M));
+ SCHAR refNrg_e = (SCHAR)((LONG)(*pIenv) & MASK_E) - NRG_EXP_OFFSET;
+
+ UCHAR sinePresentFlag = 0;
+ int li = table[j];
+ int ui = table[j+1];
+
+ for (k=li; k<ui; k++) {
+ sinePresentFlag |= (i >= sineMapped[cc]);
+ cc++;
+ }
+
+ for (k=li; k<ui; k++) {
+ if (k >= *pUiNoise) {
+ tmpNoise = FX_SGL2FX_DBL((FIXP_SGL)((LONG)(*pNoiseLevels) & MASK_M));
+ tmpNoise_e = (SCHAR)((LONG)(*pNoiseLevels++) & MASK_E) - NOISE_EXP_OFFSET;
+
+ pUiNoise++;
+ }
+
+ FDK_ASSERT(k >= lowSubband);
+
+ if (useLP)
+ useAliasReduction[k-lowSubband] = !sinePresentFlag;
+
+ pNrgs->nrgSine[c] = FL2FXCONST_DBL(0.0f);
+ pNrgs->nrgSine_e[c] = 0;
+
+ calcSubbandGain(refNrg, refNrg_e, pNrgs, c,
+ tmpNoise, tmpNoise_e,
+ sinePresentFlag, i >= sineMapped[c],
+ noNoiseFlag);
+
+ pNrgs->nrgRef[c] = refNrg;
+ pNrgs->nrgRef_e[c] = refNrg_e;
+
+ c++;
+ }
+ pIenv++;
+ }
+ }
+
+ /*
+ Noise limiting
+ */
+
+ for (c = 0; c < hFreq->noLimiterBands; c++) {
+
+ FIXP_DBL sumRef, boostGain, maxGain;
+ FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
+ SCHAR sumRef_e, boostGain_e, maxGain_e, accu_e = 0;
+
+ calcAvgGain(pNrgs,
+ hFreq->limiterBandTable[c], hFreq->limiterBandTable[c+1],
+ &sumRef, &sumRef_e,
+ &maxGain, &maxGain_e);
+
+ /* Multiply maxGain with limiterGain: */
+ maxGain = fMult(maxGain, FDK_sbrDecoder_sbr_limGains_m[hHeaderData->bs_data.limiterGains]);
+ maxGain_e += FDK_sbrDecoder_sbr_limGains_e[hHeaderData->bs_data.limiterGains];
+
+ /* Scale mantissa of MaxGain into range between 0.5 and 1: */
+ if (maxGain == FL2FXCONST_DBL(0.0f))
+ maxGain_e = -FRACT_BITS;
+ else {
+ SCHAR charTemp = CountLeadingBits(maxGain);
+ maxGain_e -= charTemp;
+ maxGain <<= (int)charTemp;
+ }
+
+ if (maxGain_e >= maxGainLimit_e) { /* upper limit (e.g. 96 dB) */
+ maxGain = FL2FXCONST_DBL(0.5f);
+ maxGain_e = maxGainLimit_e;
+ }
+
+
+ /* Every subband gain is compared to the scaled "average gain"
+ and limited if necessary: */
+ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c+1]; k++) {
+ if ( (pNrgs->nrgGain_e[k] > maxGain_e) || (pNrgs->nrgGain_e[k] == maxGain_e && pNrgs->nrgGain[k]>maxGain) ) {
+
+ FIXP_DBL noiseAmp;
+ SCHAR noiseAmp_e;
+
+ FDK_divide_MantExp(maxGain, maxGain_e, pNrgs->nrgGain[k], pNrgs->nrgGain_e[k], &noiseAmp, &noiseAmp_e);
+ pNrgs->noiseLevel[k] = fMult(pNrgs->noiseLevel[k],noiseAmp);
+ pNrgs->noiseLevel_e[k] += noiseAmp_e;
+ pNrgs->nrgGain[k] = maxGain;
+ pNrgs->nrgGain_e[k] = maxGain_e;
+ }
+ }
+
+ /* -- Boost gain
+ Calculate and apply boost factor for each limiter band:
+ 1. Check how much energy would be present when using the limited gain
+ 2. Calculate boost factor by comparison with reference energy
+ 3. Apply boost factor to compensate for the energy loss due to limiting
+ */
+ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {
+
+ /* 1.a Add energy of adjusted signal (using preliminary gain) */
+ FIXP_DBL tmp = fMult(pNrgs->nrgGain[k],pNrgs->nrgEst[k]);
+ SCHAR tmp_e = pNrgs->nrgGain_e[k] + pNrgs->nrgEst_e[k];
+ FDK_add_MantExp(tmp, tmp_e, accu, accu_e, &accu, &accu_e);
+
+ /* 1.b Add sine energy (if present) */
+ if(pNrgs->nrgSine[k] != FL2FXCONST_DBL(0.0f)) {
+ FDK_add_MantExp(pNrgs->nrgSine[k], pNrgs->nrgSine_e[k], accu, accu_e, &accu, &accu_e);
+ }
+ else {
+ /* 1.c Add noise energy (if present) */
+ if(noNoiseFlag == 0) {
+ FDK_add_MantExp(pNrgs->noiseLevel[k], pNrgs->noiseLevel_e[k], accu, accu_e, &accu, &accu_e);
+ }
+ }
+ }
+
+ /* 2.a Calculate ratio of wanted energy and accumulated energy */
+ if (accu == (FIXP_DBL)0) { /* If divisor is 0, limit quotient to +4 dB */
+ boostGain = FL2FXCONST_DBL(0.6279716f);
+ boostGain_e = 2;
+ } else {
+ INT div_e;
+ boostGain = fDivNorm(sumRef, accu, &div_e);
+ boostGain_e = sumRef_e - accu_e + div_e;
+ }
+
+
+ /* 2.b Result too high? --> Limit the boost factor to +4 dB */
+ if((boostGain_e > 3) ||
+ (boostGain_e == 2 && boostGain > FL2FXCONST_DBL(0.6279716f)) ||
+ (boostGain_e == 3 && boostGain > FL2FXCONST_DBL(0.3139858f)) )
+ {
+ boostGain = FL2FXCONST_DBL(0.6279716f);
+ boostGain_e = 2;
+ }
+ /* 3. Multiply all signal components with the boost factor */
+ for (k = hFreq->limiterBandTable[c]; k < hFreq->limiterBandTable[c + 1]; k++) {
+ pNrgs->nrgGain[k] = fMultDiv2(pNrgs->nrgGain[k],boostGain);
+ pNrgs->nrgGain_e[k] = pNrgs->nrgGain_e[k] + boostGain_e + 1;
+
+ pNrgs->nrgSine[k] = fMultDiv2(pNrgs->nrgSine[k],boostGain);
+ pNrgs->nrgSine_e[k] = pNrgs->nrgSine_e[k] + boostGain_e + 1;
+
+ pNrgs->noiseLevel[k] = fMultDiv2(pNrgs->noiseLevel[k],boostGain);
+ pNrgs->noiseLevel_e[k] = pNrgs->noiseLevel_e[k] + boostGain_e + 1;
+ }
+ }
+ /* End of noise limiting */
+
+ if (useLP)
+ aliasingReduction(degreeAlias+lowSubband,
+ pNrgs,
+ useAliasReduction,
+ noSubbands);
+
+ /* For the timeslots within the range for the output frame,
+ use the same scale for the noise levels.
+ Drawback: If the envelope exceeds the frame border, the noise levels
+ will have to be rescaled later to fit final_e of
+ the gain-values.
+ */
+ noise_e = (start_pos < no_cols) ? adj_e : final_e;
+
+ /*
+ Convert energies to amplitude levels
+ */
+ for (k=0; k<noSubbands; k++) {
+ FDK_sqrt_MantExp(&pNrgs->nrgSine[k], &pNrgs->nrgSine_e[k], &noise_e);
+ FDK_sqrt_MantExp(&pNrgs->nrgGain[k], &pNrgs->nrgGain_e[k], &pNrgs->nrgGain_e[k]);
+ FDK_sqrt_MantExp(&pNrgs->noiseLevel[k], &pNrgs->noiseLevel_e[k], &noise_e);
+ }
+
+
+
+ /*
+ Apply calculated gains and adaptive noise
+ */
+
+ /* assembleHfSignals() */
+ {
+ int scale_change, sc_change;
+ FIXP_SGL smooth_ratio;
+ int filtBufferNoiseShift=0;
+
+ /* Initialize smoothing buffers with the first valid values */
+ if (h_sbr_cal_env->startUp)
+ {
+ if (!useLP) {
+ h_sbr_cal_env->filtBufferNoise_e = noise_e;
+
+ FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR));
+ FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
+ FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL));
+
+ }
+ h_sbr_cal_env->startUp = 0;
+ }
+
+ if (!useLP) {
+
+ equalizeFiltBufferExp(h_sbr_cal_env->filtBuffer, /* buffered */
+ h_sbr_cal_env->filtBuffer_e, /* buffered */
+ pNrgs->nrgGain, /* current */
+ pNrgs->nrgGain_e, /* current */
+ noSubbands);
+
+ /* Adapt exponent of buffered noise levels to the current exponent
+ so they can easily be smoothed */
+ if((h_sbr_cal_env->filtBufferNoise_e - noise_e)>=0) {
+ int shift = fixMin(DFRACT_BITS-1,(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
+ for (k=0; k<noSubbands; k++)
+ h_sbr_cal_env->filtBufferNoise[k] <<= shift;
+ }
+ else {
+ int shift = fixMin(DFRACT_BITS-1,-(int)(h_sbr_cal_env->filtBufferNoise_e - noise_e));
+ for (k=0; k<noSubbands; k++)
+ h_sbr_cal_env->filtBufferNoise[k] >>= shift;
+ }
+
+ h_sbr_cal_env->filtBufferNoise_e = noise_e;
+ }
+
+ /* find best scaling! */
+ scale_change = -(DFRACT_BITS-1);
+ for(k=0;k<noSubbands;k++) {
+ scale_change = fixMax(scale_change,(int)pNrgs->nrgGain_e[k]);
+ }
+ sc_change = (start_pos<no_cols)? adj_e - input_e : final_e - input_e;
+
+ if ((scale_change-sc_change+1)<0)
+ scale_change-=(scale_change-sc_change+1);
+
+ scale_change = (scale_change-sc_change)+1;
+
+ for(k=0;k<noSubbands;k++) {
+ int sc = scale_change-pNrgs->nrgGain_e[k] + (sc_change-1);
+ pNrgs->nrgGain[k] >>= sc;
+ pNrgs->nrgGain_e[k] += sc;
+ }
+
+ if (!useLP) {
+ for(k=0;k<noSubbands;k++) {
+ int sc = scale_change-h_sbr_cal_env->filtBuffer_e[k] + (sc_change-1);
+ h_sbr_cal_env->filtBuffer[k] >>= sc;
+ }
+ }
+
+ for (j = start_pos; j < stop_pos; j++)
+ {
+ /* This timeslot is located within the first part of the processing buffer
+ and will be fed into the QMF-synthesis for the current frame.
+ adj_e - input_e
+ This timeslot will not yet be fed into the QMF so we do not care
+ about the adj_e.
+ sc_change = final_e - input_e
+ */
+ if ( (j==no_cols) && (start_pos<no_cols) )
+ {
+ int shift = (int) (noise_e - final_e);
+ if (!useLP)
+ filtBufferNoiseShift = shift; /* shifting of h_sbr_cal_env->filtBufferNoise[k] will be applied in function adjustTimeSlotHQ() */
+ if (shift>=0) {
+ shift = fixMin(DFRACT_BITS-1,shift);
+ for (k=0; k<noSubbands; k++) {
+ pNrgs->nrgSine[k] <<= shift;
+ pNrgs->noiseLevel[k] <<= shift;
+ /*
+ if (!useLP)
+ h_sbr_cal_env->filtBufferNoise[k] <<= shift;
+ */
+ }
+ }
+ else {
+ shift = fixMin(DFRACT_BITS-1,-shift);
+ for (k=0; k<noSubbands; k++) {
+ pNrgs->nrgSine[k] >>= shift;
+ pNrgs->noiseLevel[k] >>= shift;
+ /*
+ if (!useLP)
+ h_sbr_cal_env->filtBufferNoise[k] >>= shift;
+ */
+ }
+ }
+
+ /* update noise scaling */
+ noise_e = final_e;
+ if (!useLP)
+ h_sbr_cal_env->filtBufferNoise_e = noise_e; /* scaling value unused! */
+
+ /* update gain buffer*/
+ sc_change -= (final_e - input_e);
+
+ if (sc_change<0) {
+ for(k=0;k<noSubbands;k++) {
+ pNrgs->nrgGain[k] >>= -sc_change;
+ pNrgs->nrgGain_e[k] += -sc_change;
+ }
+ if (!useLP) {
+ for(k=0;k<noSubbands;k++) {
+ h_sbr_cal_env->filtBuffer[k] >>= -sc_change;
+ }
+ }
+ } else {
+ scale_change+=sc_change;
+ }
+
+ } // if
+
+ if (!useLP) {
+
+ /* Prevent the smoothing filter from running on constant levels */
+ if (j-start_pos < smooth_length)
+ smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];
+
+ else
+ smooth_ratio = FL2FXCONST_SGL(0.0f);
+
+ adjustTimeSlotHQ(&analysBufferReal[j][lowSubband],
+ &analysBufferImag[j][lowSubband],
+ h_sbr_cal_env,
+ pNrgs,
+ lowSubband,
+ noSubbands,
+ scale_change,
+ smooth_ratio,
+ noNoiseFlag,
+ filtBufferNoiseShift);
+ }
+ else
+ {
+ adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+ pNrgs,
+ &h_sbr_cal_env->harmIndex,
+ lowSubband,
+ noSubbands,
+ scale_change,
+ noNoiseFlag,
+ &h_sbr_cal_env->phaseIndex,
+ (flags & SBRDEC_ELD_GRID));
+ }
+ } // for
+
+ if (!useLP) {
+ /* Update time-smoothing-buffers for gains and noise levels
+ The gains and the noise values of the current envelope are copied into the buffer.
+ This has to be done at the end of each envelope as the values are required for
+ a smooth transition to the next envelope. */
+ FDKmemcpy(h_sbr_cal_env->filtBuffer, pNrgs->nrgGain, noSubbands*sizeof(FIXP_DBL));
+ FDKmemcpy(h_sbr_cal_env->filtBuffer_e, pNrgs->nrgGain_e, noSubbands*sizeof(SCHAR));
+ FDKmemcpy(h_sbr_cal_env->filtBufferNoise, pNrgs->noiseLevel, noSubbands*sizeof(FIXP_DBL));
+ }
+
+ }
+ C_ALLOC_SCRATCH_END(pNrgs, ENV_CALC_NRGS, 1);
+ }
+
+ /* Rescale output samples */
+ {
+ FIXP_DBL maxVal;
+ int ov_reserve, reserve;
+
+ /* Determine headroom in old adjusted samples */
+ maxVal = maxSubbandSample( analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ lowSubband,
+ highSubband,
+ 0,
+ first_start);
+
+ ov_reserve = fNorm(maxVal);
+
+ /* Determine headroom in new adjusted samples */
+ maxVal = maxSubbandSample( analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ lowSubband,
+ highSubband,
+ first_start,
+ no_cols);
+
+ reserve = fNorm(maxVal);
+
+ /* Determine common output exponent */
+ if (ov_adj_e - ov_reserve > adj_e - reserve ) /* set output_e to the maximum */
+ output_e = ov_adj_e - ov_reserve;
+ else
+ output_e = adj_e - reserve;
+
+ /* Rescale old samples */
+ rescaleSubbandSamples( analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ lowSubband, highSubband,
+ 0, first_start,
+ ov_adj_e - output_e);
+
+ /* Rescale new samples */
+ rescaleSubbandSamples( analysBufferReal,
+ (useLP) ? NULL : analysBufferImag,
+ lowSubband, highSubband,
+ first_start, no_cols,
+ adj_e - output_e);
+ }
+
+ /* Update hb_scale */
+ sbrScaleFactor->hb_scale = EXP2SCALE(output_e);
+
+ /* Save the current final exponent for the next frame: */
+ sbrScaleFactor->ov_hb_scale = EXP2SCALE(final_e);
+
+
+ /* We need to remeber to the next frame that the transient
+ will occur in the first envelope (if tranEnv == nEnvelopes). */
+ if(hFrameData->frameInfo.tranEnv == hFrameData->frameInfo.nEnvelopes)
+ h_sbr_cal_env->prevTranEnv = 0;
+ else
+ h_sbr_cal_env->prevTranEnv = -1;
+
+}
+
+
+/*!
+ \brief Create envelope instance
+
+ Must be called once for each channel before calculateSbrEnvelope() can be used.
+
+ \return errorCode, 0 if successful
+*/
+SBR_ERROR
+createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs, /*!< pointer to envelope instance */
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< static SBR control data, initialized with defaults */
+ const int chan, /*!< Channel for which to assign buffers */
+ const UINT flags)
+{
+ SBR_ERROR err = SBRDEC_OK;
+ int i;
+
+ /* Clear previous missing harmonics flags */
+ for (i=0; i<(MAX_FREQ_COEFFS+15)>>4; i++) {
+ hs->harmFlagsPrev[i] = 0;
+ }
+ hs->harmIndex = 0;
+
+ /*
+ Setup pointers for time smoothing.
+ The buffer itself will be initialized later triggered by the startUp-flag.
+ */
+ hs->prevTranEnv = -1;
+
+
+ /* initialization */
+ resetSbrEnvelopeCalc(hs);
+
+ if (chan==0) { /* do this only once */
+ err = resetFreqBandTables(hHeaderData, flags);
+ }
+
+ return err;
+}
+
+/*!
+ \brief Create envelope instance
+
+ Must be called once for each channel before calculateSbrEnvelope() can be used.
+
+ \return errorCode, 0 if successful
+*/
+int
+deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hs)
+{
+ return 0;
+}
+
+
+/*!
+ \brief Reset envelope instance
+
+ This function must be called for each channel on a change of configuration.
+ Note that resetFreqBandTables should also be called in this case.
+
+ \return errorCode, 0 if successful
+*/
+void
+resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to envelope instance */
+{
+ hCalEnv->phaseIndex = 0;
+
+ /* Noise exponent needs to be reset because the output exponent for the next frame depends on it */
+ hCalEnv->filtBufferNoise_e = 0;
+
+ hCalEnv->startUp = 1;
+}
+
+
+/*!
+ \brief Equalize exponents of the buffered gain values and the new ones
+
+ After equalization of exponents, the FIR-filter addition for smoothing
+ can be performed.
+ This function is called once for each envelope before adjusting.
+*/
+/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer, /*!< bufferd gains */
+ SCHAR *filtBuffer_e, /*!< exponents of bufferd gains */
+ FIXP_DBL *nrgGain, /*!< gains for current envelope */
+ SCHAR *nrgGain_e, /*!< exponents of gains for current envelope */
+ int subbands) /*!< Number of QMF subbands */
+{
+ int band;
+ int diff;
+
+ for (band=0; band<subbands; band++){
+ diff = (int) (nrgGain_e[band] - filtBuffer_e[band]);
+ if (diff>0) {
+ filtBuffer[band] >>= diff; /* Compensate for the scale change by shifting the mantissa. */
+ filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */
+ }
+ else if (diff<0) {
+ /* The buffered gains seem to be larger, but maybe there
+ are some unused bits left in the mantissa */
+
+ int reserve = CntLeadingZeros(fixp_abs(filtBuffer[band]))-1;
+
+ if ((-diff) <= reserve) {
+ /* There is enough space in the buffered mantissa so
+ that we can take the new exponent as common.
+ */
+ filtBuffer[band] <<= (-diff);
+ filtBuffer_e[band] += diff; /* becomes equal to *ptrNewExp */
+ }
+ else {
+ filtBuffer[band] <<= reserve; /* Shift the mantissa as far as possible: */
+ filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */
+
+ /* For the remaining difference, change the new gain value */
+ diff = fixMin(-(reserve + diff),DFRACT_BITS-1);
+ nrgGain[band] >>= diff;
+ nrgGain_e[band] += diff;
+ }
+ }
+ }
+}
+
+/*!
+ \brief Shift left the mantissas of all subband samples
+ in the giventime and frequency range by the specified number of bits.
+
+ This function is used to rescale the audio data in the overlap buffer
+ which has already been envelope adjusted with the last frame.
+*/
+void rescaleSubbandSamples(FIXP_DBL ** re, /*!< Real part of input and output subband samples */
+ FIXP_DBL ** im, /*!< Imaginary part of input and output subband samples */
+ int lowSubband, /*!< Begin of frequency range to process */
+ int highSubband, /*!< End of frequency range to process */
+ int start_pos, /*!< Begin of time rage (QMF-timeslot) */
+ int next_pos, /*!< End of time rage (QMF-timeslot) */
+ int shift) /*!< number of bits to shift */
+{
+ int width = highSubband-lowSubband;
+
+ if ( (width > 0) && (shift!=0) ) {
+ if (im!=NULL) {
+ for (int l=start_pos; l<next_pos; l++) {
+ scaleValues(&re[l][lowSubband], width, shift);
+ scaleValues(&im[l][lowSubband], width, shift);
+ }
+ } else
+ {
+ for (int l=start_pos; l<next_pos; l++) {
+ scaleValues(&re[l][lowSubband], width, shift);
+ }
+ }
+ }
+}
+
+
+/*!
+ \brief Determine headroom for shifting
+
+ Determine by how much the spectrum can be shifted left
+ for better accuracy in later processing.
+
+ \return Number of free bits in the biggest spectral value
+*/
+
+FIXP_DBL maxSubbandSample( FIXP_DBL ** re, /*!< Real part of input and output subband samples */
+ FIXP_DBL ** im, /*!< Real part of input and output subband samples */
+ int lowSubband, /*!< Begin of frequency range to process */
+ int highSubband, /*!< Number of QMF bands to process */
+ int start_pos, /*!< Begin of time rage (QMF-timeslot) */
+ int next_pos /*!< End of time rage (QMF-timeslot) */
+ )
+{
+ FIXP_DBL maxVal = FL2FX_DBL(0.0f);
+ unsigned int width = highSubband - lowSubband;
+
+ FDK_ASSERT(width <= (64));
+
+ if ( width > 0 ) {
+ if (im!=NULL)
+ {
+ for (int l=start_pos; l<next_pos; l++)
+ {
+#ifdef FUNCTION_FDK_get_maxval
+ maxVal = FDK_get_maxval(maxVal, &re[l][lowSubband], &im[l][lowSubband], width);
+#else
+ int k=width;
+ FIXP_DBL *reTmp = &re[l][lowSubband];
+ FIXP_DBL *imTmp = &im[l][lowSubband];
+ do{
+ FIXP_DBL tmp1 = *(reTmp++);
+ FIXP_DBL tmp2 = *(imTmp++);
+ maxVal |= (FIXP_DBL)((LONG)(tmp1)^((LONG)tmp1>>(DFRACT_BITS-1)));
+ maxVal |= (FIXP_DBL)((LONG)(tmp2)^((LONG)tmp2>>(DFRACT_BITS-1)));
+ } while(--k!=0);
+#endif
+ }
+ } else
+ {
+ for (int l=start_pos; l<next_pos; l++) {
+ int k=width;
+ FIXP_DBL *reTmp = &re[l][lowSubband];
+ do{
+ FIXP_DBL tmp = *(reTmp++);
+ maxVal |= (FIXP_DBL)((LONG)(tmp)^((LONG)tmp>>(DFRACT_BITS-1)));
+ }while(--k!=0);
+ }
+ }
+ }
+
+ return(maxVal);
+}
+
+#define SHIFT_BEFORE_SQUARE (3) /* (7/2) */
+/*!<
+ If the accumulator does not provide enough overflow bits or
+ does not provide a high dynamic range, the below energy calculation
+ requires an additional shift operation for each sample.
+ On the other hand, doing the shift allows using a single-precision
+ multiplication for the square (at least 16bit x 16bit).
+ For even values of OVRFLW_BITS (0, 2, 4, 6), saturated arithmetic
+ is required for the energy accumulation.
+ Theoretically, the sample-squares can sum up to a value of 76,
+ requiring 7 overflow bits. However since such situations are *very*
+ rare, accu can be limited to 64.
+ In case native saturated arithmetic is not available, overflows
+ can be prevented by replacing the above #define by
+ #define SHIFT_BEFORE_SQUARE ((8 - OVRFLW_BITS) / 2)
+ which will result in slightly reduced accuracy.
+*/
+
+/*!
+ \brief Estimates the mean energy of each filter-bank channel for the
+ duration of the current envelope
+
+ This function is used when interpolFreq is true.
+*/
+/*static*/ void calcNrgPerSubband(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
+ FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */
+ int lowSubband, /*!< Begin of the SBR frequency range */
+ int highSubband, /*!< High end of the SBR frequency range */
+ int start_pos, /*!< First QMF-slot of current envelope */
+ int next_pos, /*!< Last QMF-slot of current envelope + 1 */
+ SCHAR frameExp, /*!< Common exponent for all input samples */
+ FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */
+ SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */
+{
+ FIXP_SGL invWidth;
+ SCHAR preShift;
+ SCHAR shift;
+ FIXP_DBL sum;
+ int k,l;
+
+ /* Divide by width of envelope later: */
+ invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
+ /* The common exponent needs to be doubled because all mantissas are squared: */
+ frameExp = frameExp << 1;
+
+ for (k=lowSubband; k<highSubband; k++) {
+ FIXP_DBL bufferReal[(((1024)/(32))+(6))];
+ FIXP_DBL bufferImag[(((1024)/(32))+(6))];
+ FIXP_DBL maxVal = FL2FX_DBL(0.0f);
+
+ if (analysBufferImag!=NULL)
+ {
+ for (l=start_pos;l<next_pos;l++)
+ {
+ bufferImag[l] = analysBufferImag[l][k];
+ maxVal |= (FIXP_DBL)((LONG)(bufferImag[l])^((LONG)bufferImag[l]>>(DFRACT_BITS-1)));
+ bufferReal[l] = analysBufferReal[l][k];
+ maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
+ }
+ }
+ else
+ {
+ for (l=start_pos;l<next_pos;l++)
+ {
+ bufferReal[l] = analysBufferReal[l][k];
+ maxVal |= (FIXP_DBL)((LONG)(bufferReal[l])^((LONG)bufferReal[l]>>(DFRACT_BITS-1)));
+ }
+ }
+
+ if (maxVal!=FL2FXCONST_DBL(0.f)) {
+
+
+ /* If the accu does not provide enough overflow bits, we cannot
+ shift the samples up to the limit.
+ Instead, keep up to 3 free bits in each sample, i.e. up to
+ 6 bits after calculation of square.
+ Please note the comment on saturated arithmetic above!
+ */
+ FIXP_DBL accu = FL2FXCONST_DBL(0.0f);
+ preShift = CntLeadingZeros(maxVal)-1;
+ preShift -= SHIFT_BEFORE_SQUARE;
+
+ if (preShift>=0) {
+ if (analysBufferImag!=NULL) {
+ for (l=start_pos; l<next_pos; l++) {
+ FIXP_DBL temp1 = bufferReal[l] << (int)preShift;
+ FIXP_DBL temp2 = bufferImag[l] << (int)preShift;
+ accu = fPow2AddDiv2(accu, temp1);
+ accu = fPow2AddDiv2(accu, temp2);
+ }
+ } else
+ {
+ for (l=start_pos; l<next_pos; l++) {
+ FIXP_DBL temp = bufferReal[l] << (int)preShift;
+ accu = fPow2AddDiv2(accu, temp);
+ }
+ }
+ }
+ else { /* if negative shift value */
+ int negpreShift = -preShift;
+ if (analysBufferImag!=NULL) {
+ for (l=start_pos; l<next_pos; l++) {
+ FIXP_DBL temp1 = bufferReal[l] >> (int)negpreShift;
+ FIXP_DBL temp2 = bufferImag[l] >> (int)negpreShift;
+ accu = fPow2AddDiv2(accu, temp1);
+ accu = fPow2AddDiv2(accu, temp2);
+ }
+ } else
+ {
+ for (l=start_pos; l<next_pos; l++) {
+ FIXP_DBL temp = bufferReal[l] >> (int)negpreShift;
+ accu = fPow2AddDiv2(accu, temp);
+ }
+ }
+ }
+ accu <<= 1;
+
+ /* Convert double precision to Mantissa/Exponent: */
+ shift = fNorm(accu);
+ sum = accu << (int)shift;
+
+ /* Divide by width of envelope and apply frame scale: */
+ *nrgEst++ = fMult(sum, invWidth);
+ shift += 2 * preShift;
+ if (analysBufferImag!=NULL)
+ *nrgEst_e++ = frameExp - shift;
+ else
+ *nrgEst_e++ = frameExp - shift + 1; /* +1 due to missing imag. part */
+ } /* maxVal!=0 */
+ else {
+
+ /* Prevent a zero-mantissa-number from being misinterpreted
+ due to its exponent. */
+ *nrgEst++ = FL2FXCONST_DBL(0.0f);
+ *nrgEst_e++ = 0;
+ }
+ }
+}
+
+/*!
+ \brief Estimates the mean energy of each Scale factor band for the
+ duration of the current envelope.
+
+ This function is used when interpolFreq is false.
+*/
+/*static*/ void calcNrgPerSfb(FIXP_DBL **analysBufferReal, /*!< Real part of subband samples */
+ FIXP_DBL **analysBufferImag, /*!< Imaginary part of subband samples */
+ int nSfb, /*!< Number of scale factor bands */
+ UCHAR *freqBandTable, /*!< First Subband for each Sfb */
+ int start_pos, /*!< First QMF-slot of current envelope */
+ int next_pos, /*!< Last QMF-slot of current envelope + 1 */
+ SCHAR input_e, /*!< Common exponent for all input samples */
+ FIXP_DBL *nrgEst, /*!< resulting Energy (0..1) */
+ SCHAR *nrgEst_e ) /*!< Exponent of resulting Energy */
+{
+ FIXP_SGL invWidth;
+ FIXP_DBL temp;
+ SCHAR preShift;
+ SCHAR shift, sum_e;
+ FIXP_DBL sum;
+
+ int j,k,l,li,ui;
+ FIXP_DBL sumAll, sumLine; /* Single precision would be sufficient,
+ but overflow bits are required for accumulation */
+
+ /* Divide by width of envelope later: */
+ invWidth = FX_DBL2FX_SGL(GetInvInt(next_pos - start_pos));
+ /* The common exponent needs to be doubled because all mantissas are squared: */
+ input_e = input_e << 1;
+
+ for(j=0; j<nSfb; j++) {
+ li = freqBandTable[j];
+ ui = freqBandTable[j+1];
+
+ FIXP_DBL maxVal = maxSubbandSample( analysBufferReal,
+ analysBufferImag,
+ li,
+ ui,
+ start_pos,
+ next_pos );
+
+ if (maxVal!=FL2FXCONST_DBL(0.f)) {
+
+ preShift = CntLeadingZeros(maxVal)-1;
+
+ /* If the accu does not provide enough overflow bits, we cannot
+ shift the samples up to the limit.
+ Instead, keep up to 3 free bits in each sample, i.e. up to
+ 6 bits after calculation of square.
+ Please note the comment on saturated arithmetic above!
+ */
+ preShift -= SHIFT_BEFORE_SQUARE;
+
+ sumAll = FL2FXCONST_DBL(0.0f);
+
+
+ for (k=li; k<ui; k++) {
+
+ sumLine = FL2FXCONST_DBL(0.0f);
+
+ if (analysBufferImag!=NULL) {
+ if (preShift>=0) {
+ for (l=start_pos; l<next_pos; l++) {
+ temp = analysBufferReal[l][k] << (int)preShift;
+ sumLine += fPow2Div2(temp);
+ temp = analysBufferImag[l][k] << (int)preShift;
+ sumLine += fPow2Div2(temp);
+
+ }
+ } else {
+ for (l=start_pos; l<next_pos; l++) {
+ temp = analysBufferReal[l][k] >> -(int)preShift;
+ sumLine += fPow2Div2(temp);
+ temp = analysBufferImag[l][k] >> -(int)preShift;
+ sumLine += fPow2Div2(temp);
+ }
+ }
+ } else
+ {
+ if (preShift>=0) {
+ for (l=start_pos; l<next_pos; l++) {
+ temp = analysBufferReal[l][k] << (int)preShift;
+ sumLine += fPow2Div2(temp);
+ }
+ } else {
+ for (l=start_pos; l<next_pos; l++) {
+ temp = analysBufferReal[l][k] >> -(int)preShift;
+ sumLine += fPow2Div2(temp);
+ }
+ }
+ }
+
+ /* The number of QMF-channels per SBR bands may be up to 15.
+ Shift right to avoid overflows in sum over all channels. */
+ sumLine = sumLine >> (4-1);
+ sumAll += sumLine;
+ }
+
+ /* Convert double precision to Mantissa/Exponent: */
+ shift = fNorm(sumAll);
+ sum = sumAll << (int)shift;
+
+ /* Divide by width of envelope: */
+ sum = fMult(sum,invWidth);
+
+ /* Divide by width of Sfb: */
+ sum = fMult(sum, FX_DBL2FX_SGL(GetInvInt(ui-li)));
+
+ /* Set all Subband energies in the Sfb to the average energy: */
+ if (analysBufferImag!=NULL)
+ sum_e = input_e + 4 - shift; /* -4 to compensate right-shift */
+ else
+ sum_e = input_e + 4 + 1 - shift; /* -4 to compensate right-shift; +1 due to missing imag. part */
+
+ sum_e -= 2 * preShift;
+ } /* maxVal!=0 */
+ else {
+
+ /* Prevent a zero-mantissa-number from being misinterpreted
+ due to its exponent. */
+ sum = FL2FXCONST_DBL(0.0f);
+ sum_e = 0;
+ }
+
+ for (k=li; k<ui; k++)
+ {
+ *nrgEst++ = sum;
+ *nrgEst_e++ = sum_e;
+ }
+ }
+}
+
+
+/*!
+ \brief Calculate gain, noise, and additional sine level for one subband.
+
+ The resulting energy gain is given by mantissa and exponent.
+*/
+/*static*/ void calcSubbandGain(FIXP_DBL nrgRef, /*!< Reference Energy according to envelope data */
+ SCHAR nrgRef_e, /*!< Reference Energy according to envelope data (exponent) */
+ ENV_CALC_NRGS* nrgs,
+ int i,
+ FIXP_DBL tmpNoise, /*!< Relative noise level */
+ SCHAR tmpNoise_e, /*!< Relative noise level (exponent) */
+ UCHAR sinePresentFlag, /*!< Indicates if sine is present on band */
+ UCHAR sineMapped, /*!< Indicates if sine must be added */
+ int noNoiseFlag) /*!< Flag to suppress noise addition */
+{
+ FIXP_DBL nrgEst = nrgs->nrgEst[i]; /*!< Energy in transposed signal */
+ SCHAR nrgEst_e = nrgs->nrgEst_e[i]; /*!< Energy in transposed signal (exponent) */
+ FIXP_DBL *ptrNrgGain = &nrgs->nrgGain[i]; /*!< Resulting energy gain */
+ SCHAR *ptrNrgGain_e = &nrgs->nrgGain_e[i]; /*!< Resulting energy gain (exponent) */
+ FIXP_DBL *ptrNoiseLevel = &nrgs->noiseLevel[i]; /*!< Resulting absolute noise energy */
+ SCHAR *ptrNoiseLevel_e = &nrgs->noiseLevel_e[i]; /*!< Resulting absolute noise energy (exponent) */
+ FIXP_DBL *ptrNrgSine = &nrgs->nrgSine[i]; /*!< Additional sine energy */
+ SCHAR *ptrNrgSine_e = &nrgs->nrgSine_e[i]; /*!< Additional sine energy (exponent) */
+
+ FIXP_DBL a, b, c;
+ SCHAR a_e, b_e, c_e;
+
+ /*
+ This addition of 1 prevents divisions by zero in the reference code.
+ For very small energies in nrgEst, it prevents the gains from becoming
+ very high which could cause some trouble due to the smoothing.
+ */
+ b_e = (int)(nrgEst_e - 1);
+ if (b_e>=0) {
+ nrgEst = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (nrgEst >> 1);
+ nrgEst_e += 1; /* shift by 1 bit to avoid overflow */
+
+ } else {
+ nrgEst = (nrgEst >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
+ nrgEst_e = 2; /* shift by 1 bit to avoid overflow */
+ }
+
+ /* A = NrgRef * TmpNoise */
+ a = fMult(nrgRef,tmpNoise);
+ a_e = nrgRef_e + tmpNoise_e;
+
+ /* B = 1 + TmpNoise */
+ b_e = (int)(tmpNoise_e - 1);
+ if (b_e>=0) {
+ b = (FL2FXCONST_DBL(0.5f) >> (INT)fixMin(b_e+1,DFRACT_BITS-1)) + (tmpNoise >> 1);
+ b_e = tmpNoise_e + 1; /* shift by 1 bit to avoid overflow */
+ } else {
+ b = (tmpNoise >> (INT)(fixMin(-b_e+1,DFRACT_BITS-1))) + (FL2FXCONST_DBL(0.5f) >> 1);
+ b_e = 2; /* shift by 1 bit to avoid overflow */
+ }
+
+ /* noiseLevel = A / B = (NrgRef * TmpNoise) / (1 + TmpNoise) */
+ FDK_divide_MantExp( a, a_e,
+ b, b_e,
+ ptrNoiseLevel, ptrNoiseLevel_e);
+
+ if (sinePresentFlag) {
+
+ /* C = (1 + TmpNoise) * NrgEst */
+ c = fMult(b,nrgEst);
+ c_e = b_e + nrgEst_e;
+
+ /* gain = A / C = (NrgRef * TmpNoise) / (1 + TmpNoise) * NrgEst */
+ FDK_divide_MantExp( a, a_e,
+ c, c_e,
+ ptrNrgGain, ptrNrgGain_e);
+
+ if (sineMapped) {
+
+ /* sineLevel = nrgRef/ (1 + TmpNoise) */
+ FDK_divide_MantExp( nrgRef, nrgRef_e,
+ b, b_e,
+ ptrNrgSine, ptrNrgSine_e);
+ }
+ }
+ else {
+ if (noNoiseFlag) {
+ /* B = NrgEst */
+ b = nrgEst;
+ b_e = nrgEst_e;
+ }
+ else {
+ /* B = NrgEst * (1 + TmpNoise) */
+ b = fMult(b,nrgEst);
+ b_e = b_e + nrgEst_e;
+ }
+
+
+ /* gain = nrgRef / B */
+ FDK_divide_MantExp( nrgRef, nrgRef_e,
+ b, b_e,
+ ptrNrgGain, ptrNrgGain_e);
+ }
+}
+
+
+/*!
+ \brief Calculate "average gain" for the specified subband range.
+
+ This is rather a gain of the average magnitude than the average
+ of gains!
+ The result is used as a relative limit for all gains within the
+ current "limiter band" (a certain frequency range).
+*/
+/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+ int lowSubband, /*!< Begin of the limiter band */
+ int highSubband, /*!< High end of the limiter band */
+ FIXP_DBL *ptrSumRef,
+ SCHAR *ptrSumRef_e,
+ FIXP_DBL *ptrAvgGain, /*!< Resulting overall gain (mantissa) */
+ SCHAR *ptrAvgGain_e) /*!< Resulting overall gain (exponent) */
+{
+ FIXP_DBL *nrgRef = nrgs->nrgRef; /*!< Reference Energy according to envelope data */
+ SCHAR *nrgRef_e = nrgs->nrgRef_e; /*!< Reference Energy according to envelope data (exponent) */
+ FIXP_DBL *nrgEst = nrgs->nrgEst; /*!< Energy in transposed signal */
+ SCHAR *nrgEst_e = nrgs->nrgEst_e; /*!< Energy in transposed signal (exponent) */
+
+ FIXP_DBL sumRef = 1;
+ FIXP_DBL sumEst = 1;
+ SCHAR sumRef_e = -FRACT_BITS;
+ SCHAR sumEst_e = -FRACT_BITS;
+ int k;
+
+ for (k=lowSubband; k<highSubband; k++){
+ /* Add nrgRef[k] to sumRef: */
+ FDK_add_MantExp( sumRef, sumRef_e,
+ nrgRef[k], nrgRef_e[k],
+ &sumRef, &sumRef_e );
+
+ /* Add nrgEst[k] to sumEst: */
+ FDK_add_MantExp( sumEst, sumEst_e,
+ nrgEst[k], nrgEst_e[k],
+ &sumEst, &sumEst_e );
+ }
+
+ FDK_divide_MantExp(sumRef, sumRef_e,
+ sumEst, sumEst_e,
+ ptrAvgGain, ptrAvgGain_e);
+
+ *ptrSumRef = sumRef;
+ *ptrSumRef_e = sumRef_e;
+}
+
+
+/*!
+ \brief Amplify one timeslot of the signal with the calculated gains
+ and add the noisefloor.
+*/
+
+/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal, /*!< Subband samples to be adjusted, real part */
+ ENV_CALC_NRGS* nrgs,
+ UCHAR *ptrHarmIndex, /*!< Harmonic index */
+ int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */
+ int noSubbands, /*!< Number of QMF subbands */
+ int scale_change, /*!< Number of bits to shift adjusted samples */
+ int noNoiseFlag, /*!< Flag to suppress noise addition */
+ int *ptrPhaseIndex, /*!< Start index to random number array */
+ int fCldfb) /*!< CLDFB 80 flag */
+{
+ FIXP_DBL *pGain = nrgs->nrgGain; /*!< Gains of current envelope */
+ FIXP_DBL *pNoiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */
+ FIXP_DBL *pSineLevel = nrgs->nrgSine; /*!< Sine levels */
+
+ int k;
+ int index = *ptrPhaseIndex;
+ UCHAR harmIndex = *ptrHarmIndex;
+ UCHAR freqInvFlag = (lowSubband & 1);
+ FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev;
+ int tone_count = 0;
+ int sineSign = 1;
+
+ #define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.00815f))
+ #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f*0.16773f))
+
+ /*
+ First pass for k=0 pulled out of the loop:
+ */
+
+ index = (index + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
+
+ /*
+ The next multiplication constitutes the actual envelope adjustment
+ of the signal and should be carried out with full accuracy
+ (supplying #FRACT_BITS valid bits).
+ */
+ signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+ sineLevel = *pSineLevel++;
+ sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
+
+ if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;
+
+ else if (!noNoiseFlag)
+ /* Add noisefloor to the amplified signal */
+ signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+
+ if (fCldfb) {
+
+ if (!(harmIndex&0x1)) {
+ /* harmIndex 0,2 */
+ signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
+ *ptrReal++ = signalReal;
+ }
+ else {
+ /* harmIndex 1,3 in combination with freqInvFlag */
+ int shift = (int) (scale_change+1);
+ shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
+
+ FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );
+
+ FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);
+
+
+ /* save switch and compare operations and reduce to XOR statement */
+ if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
+ *(ptrReal-1) += tmp1;
+ signalReal -= tmp2;
+ } else {
+ *(ptrReal-1) -= tmp1;
+ signalReal += tmp2;
+ }
+ *ptrReal++ = signalReal;
+ freqInvFlag = !freqInvFlag;
+ }
+
+ } else
+ {
+ if (!(harmIndex&0x1)) {
+ /* harmIndex 0,2 */
+ signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
+ *ptrReal++ = signalReal;
+ }
+ else {
+ /* harmIndex 1,3 in combination with freqInvFlag */
+ int shift = (int) (scale_change+1);
+ shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
+
+ FIXP_DBL tmp1 = (shift>=0) ? ( fMultDiv2(C1, sineLevel) >> shift )
+ : ( fMultDiv2(C1, sineLevel) << (-shift) );
+ FIXP_DBL tmp2 = fMultDiv2(C1, sineLevelNext);
+
+
+ /* save switch and compare operations and reduce to XOR statement */
+ if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
+ *(ptrReal-1) += tmp1;
+ signalReal -= tmp2;
+ } else {
+ *(ptrReal-1) -= tmp1;
+ signalReal += tmp2;
+ }
+ *ptrReal++ = signalReal;
+ freqInvFlag = !freqInvFlag;
+ }
+ }
+
+ pNoiseLevel++;
+
+ if ( noSubbands > 2 ) {
+ if (!(harmIndex&0x1)) {
+ /* harmIndex 0,2 */
+ if(!harmIndex)
+ {
+ sineSign = 0;
+ }
+
+ for (k=noSubbands-2; k!=0; k--) {
+ FIXP_DBL sinelevel = *pSineLevel++;
+ index++;
+ if (((signalReal = (sineSign ? -sinelevel : sinelevel)) == FL2FXCONST_DBL(0.0f)) && !noNoiseFlag)
+ {
+ /* Add noisefloor to the amplified signal */
+ index &= (SBR_NF_NO_RANDOM_VAL - 1);
+ signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+ }
+
+ /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+ signalReal += fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+
+ pNoiseLevel++;
+ *ptrReal++ = signalReal;
+ } /* for ... */
+ }
+ else {
+ /* harmIndex 1,3 in combination with freqInvFlag */
+ if (harmIndex==1) freqInvFlag = !freqInvFlag;
+
+ for (k=noSubbands-2; k!=0; k--) {
+ index++;
+ /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+ signalReal = fMultDiv2(*ptrReal,*pGain++) << ((int)scale_change);
+
+ if (*pSineLevel++!=FL2FXCONST_DBL(0.0f)) tone_count++;
+ else if (!noNoiseFlag) {
+ /* Add noisefloor to the amplified signal */
+ index &= (SBR_NF_NO_RANDOM_VAL - 1);
+ signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+ }
+
+ pNoiseLevel++;
+
+ if (tone_count <= 16) {
+ FIXP_DBL addSine = fMultDiv2((pSineLevel[-2] - pSineLevel[0]), C1);
+ signalReal += (freqInvFlag) ? (-addSine) : (addSine);
+ }
+
+ *ptrReal++ = signalReal;
+ freqInvFlag = !freqInvFlag;
+ } /* for ... */
+ }
+ }
+
+ if (noSubbands > -1) {
+ index++;
+ /* The next multiplication constitutes the actual envelope adjustment of the signal. */
+ signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
+ sineLevelPrev = fMultDiv2(pSineLevel[-1],FL2FX_SGL(0.0163f));
+ sineLevel = pSineLevel[0];
+
+ if (pSineLevel[0]!=FL2FXCONST_DBL(0.0f)) tone_count++;
+ else if (!noNoiseFlag) {
+ /* Add noisefloor to the amplified signal */
+ index &= (SBR_NF_NO_RANDOM_VAL - 1);
+ signalReal = signalReal + (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
+ }
+
+ if (!(harmIndex&0x1)) {
+ /* harmIndex 0,2 */
+ *ptrReal = signalReal + ( (sineSign) ? -sineLevel : sineLevel);
+ }
+ else {
+ /* harmIndex 1,3 in combination with freqInvFlag */
+ if(tone_count <= 16){
+ if (freqInvFlag) {
+ *ptrReal++ = signalReal - sineLevelPrev;
+ if (noSubbands + lowSubband < 63)
+ *ptrReal = *ptrReal + fMultDiv2(C1, sineLevel);
+ }
+ else {
+ *ptrReal++ = signalReal + sineLevelPrev;
+ if (noSubbands + lowSubband < 63)
+ *ptrReal = *ptrReal - fMultDiv2(C1, sineLevel);
+ }
+ }
+ else *ptrReal = signalReal;
+ }
+ }
+ *ptrHarmIndex = (harmIndex + 1) & 3;
+ *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
+}
+void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal, /*!< Subband samples to be adjusted, real part */
+ FIXP_DBL *RESTRICT ptrImag, /*!< Subband samples to be adjusted, imag part */
+ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+ ENV_CALC_NRGS* nrgs,
+ int lowSubband, /*!< Lowest QMF-channel in the currently used SBR range. */
+ int noSubbands, /*!< Number of QMF subbands */
+ int scale_change, /*!< Number of bits to shift adjusted samples */
+ FIXP_SGL smooth_ratio, /*!< Impact of last envelope */
+ int noNoiseFlag, /*!< Start index to random number array */
+ int filtBufferNoiseShift) /*!< Shift factor of filtBufferNoise */
+{
+
+ FIXP_DBL *RESTRICT gain = nrgs->nrgGain; /*!< Gains of current envelope */
+ FIXP_DBL *RESTRICT noiseLevel = nrgs->noiseLevel; /*!< Noise levels of current envelope */
+ FIXP_DBL *RESTRICT pSineLevel = nrgs->nrgSine; /*!< Sine levels */
+
+ FIXP_DBL *RESTRICT filtBuffer = h_sbr_cal_env->filtBuffer; /*!< Gains of last envelope */
+ FIXP_DBL *RESTRICT filtBufferNoise = h_sbr_cal_env->filtBufferNoise; /*!< Noise levels of last envelope */
+ UCHAR *RESTRICT ptrHarmIndex =&h_sbr_cal_env->harmIndex; /*!< Harmonic index */
+ int *RESTRICT ptrPhaseIndex =&h_sbr_cal_env->phaseIndex; /*!< Start index to random number array */
+
+ int k;
+ FIXP_DBL signalReal, signalImag;
+ FIXP_DBL noiseReal, noiseImag;
+ FIXP_DBL smoothedGain, smoothedNoise;
+ FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio;
+ int index = *ptrPhaseIndex;
+ UCHAR harmIndex = *ptrHarmIndex;
+ register int freqInvFlag = (lowSubband & 1);
+ FIXP_DBL sineLevel;
+ int shift;
+
+ *ptrPhaseIndex = (index+noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1);
+ *ptrHarmIndex = (harmIndex + 1) & 3;
+
+ /*
+ Possible optimization:
+ smooth_ratio and harmIndex stay constant during the loop.
+ It might be faster to include a separate loop in each path.
+
+ the check for smooth_ratio is now outside the loop and the workload
+ of the whole function decreased by about 20 %
+ */
+
+ filtBufferNoiseShift += 1; /* due to later use of fMultDiv2 instead of fMult */
+ if (filtBufferNoiseShift<0)
+ shift = fixMin(DFRACT_BITS-1,-filtBufferNoiseShift);
+ else
+ shift = fixMin(DFRACT_BITS-1, filtBufferNoiseShift);
+
+ if (smooth_ratio > FL2FXCONST_SGL(0.0f)) {
+
+ for (k=0; k<noSubbands; k++) {
+ /*
+ Smoothing: The old envelope has been bufferd and a certain ratio
+ of the old gains and noise levels is used.
+ */
+
+ smoothedGain = fMult(smooth_ratio,filtBuffer[k]) +
+ fMult(direct_ratio,gain[k]);
+
+ if (filtBufferNoiseShift<0) {
+ smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])>>shift) +
+ fMult(direct_ratio,noiseLevel[k]);
+ }
+ else {
+ smoothedNoise = (fMultDiv2(smooth_ratio,filtBufferNoise[k])<<shift) +
+ fMult(direct_ratio,noiseLevel[k]);
+ }
+
+ /*
+ The next 2 multiplications constitute the actual envelope adjustment
+ of the signal and should be carried out with full accuracy
+ (supplying #DFRACT_BITS valid bits).
+ */
+ signalReal = fMultDiv2(*ptrReal,smoothedGain)<<((int)scale_change);
+ signalImag = fMultDiv2(*ptrImag,smoothedGain)<<((int)scale_change);
+
+ index++;
+
+ if (pSineLevel[k] != FL2FXCONST_DBL(0.0f)) {
+ sineLevel = pSineLevel[k];
+
+ switch(harmIndex) {
+ case 0:
+ *ptrReal++ = (signalReal + sineLevel);
+ *ptrImag++ = (signalImag);
+ break;
+ case 2:
+ *ptrReal++ = (signalReal - sineLevel);
+ *ptrImag++ = (signalImag);
+ break;
+ case 1:
+ *ptrReal++ = (signalReal);
+ if (freqInvFlag)
+ *ptrImag++ = (signalImag - sineLevel);
+ else
+ *ptrImag++ = (signalImag + sineLevel);
+ break;
+ case 3:
+ *ptrReal++ = signalReal;
+ if (freqInvFlag)
+ *ptrImag++ = (signalImag + sineLevel);
+ else
+ *ptrImag++ = (signalImag - sineLevel);
+ break;
+ }
+ }
+ else {
+ if (noNoiseFlag) {
+ /* Just the amplified signal is saved */
+ *ptrReal++ = (signalReal);
+ *ptrImag++ = (signalImag);
+ }
+ else {
+ /* Add noisefloor to the amplified signal */
+ index &= (SBR_NF_NO_RANDOM_VAL - 1);
+ noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise)<<4;
+ noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise)<<4;
+ *ptrReal++ = (signalReal + noiseReal);
+ *ptrImag++ = (signalImag + noiseImag);
+ }
+ }
+ freqInvFlag ^= 1;
+ }
+
+ }
+ else
+ {
+ for (k=0; k<noSubbands; k++)
+ {
+ smoothedGain = gain[k];
+ signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change;
+ signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change;
+
+ index++;
+
+ if ((sineLevel = pSineLevel[k]) != FL2FXCONST_DBL(0.0f))
+ {
+ switch (harmIndex)
+ {
+ case 0:
+ signalReal += sineLevel;
+ break;
+ case 1:
+ if (freqInvFlag)
+ signalImag -= sineLevel;
+ else
+ signalImag += sineLevel;
+ break;
+ case 2:
+ signalReal -= sineLevel;
+ break;
+ case 3:
+ if (freqInvFlag)
+ signalImag += sineLevel;
+ else
+ signalImag -= sineLevel;
+ break;
+ }
+ }
+ else
+ {
+ if (noNoiseFlag == 0)
+ {
+ /* Add noisefloor to the amplified signal */
+ smoothedNoise = noiseLevel[k];
+ index &= (SBR_NF_NO_RANDOM_VAL - 1);
+ noiseReal = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], smoothedNoise);
+ noiseImag = fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][1], smoothedNoise);
+ signalReal += noiseReal<<4;
+ signalImag += noiseImag<<4;
+ }
+ }
+ *ptrReal++ = signalReal;
+ *ptrImag++ = signalImag;
+
+ freqInvFlag ^= 1;
+ }
+ }
+}
+
+
+/*!
+ \brief Reset limiter bands.
+
+ Build frequency band table for the gain limiter dependent on
+ the previously generated transposer patch areas.
+
+ \return SBRDEC_OK if ok, SBRDEC_UNSUPPORTED_CONFIG on error
+*/
+SBR_ERROR
+ResetLimiterBands ( UCHAR *limiterBandTable, /*!< Resulting band borders in QMF channels */
+ UCHAR *noLimiterBands, /*!< Resulting number of limiter band */
+ UCHAR *freqBandTable, /*!< Table with possible band borders */
+ int noFreqBands, /*!< Number of bands in freqBandTable */
+ const PATCH_PARAM *patchParam, /*!< Transposer patch parameters */
+ int noPatches, /*!< Number of transposer patches */
+ int limiterBands) /*!< Selected 'band density' from bitstream */
+{
+ int i, k, isPatchBorder[2], loLimIndex, hiLimIndex, tempNoLim, nBands;
+ UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
+ int patchBorders[MAX_NUM_PATCHES + 1];
+ int kx, k2;
+ FIXP_DBL temp;
+
+ int lowSubband = freqBandTable[0];
+ int highSubband = freqBandTable[noFreqBands];
+
+ /* 1 limiter band. */
+ if(limiterBands == 0) {
+ limiterBandTable[0] = 0;
+ limiterBandTable[1] = highSubband - lowSubband;
+ nBands = 1;
+ } else {
+ for (i = 0; i < noPatches; i++) {
+ patchBorders[i] = patchParam[i].guardStartBand - lowSubband;
+ }
+ patchBorders[i] = highSubband - lowSubband;
+
+ /* 1.2, 2, or 3 limiter bands/octave plus bandborders at patchborders. */
+ for (k = 0; k <= noFreqBands; k++) {
+ workLimiterBandTable[k] = freqBandTable[k] - lowSubband;
+ }
+ for (k = 1; k < noPatches; k++) {
+ workLimiterBandTable[noFreqBands + k] = patchBorders[k];
+ }
+
+ tempNoLim = nBands = noFreqBands + noPatches - 1;
+ shellsort(workLimiterBandTable, tempNoLim + 1);
+
+ loLimIndex = 0;
+ hiLimIndex = 1;
+
+
+ while (hiLimIndex <= tempNoLim) {
+ k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
+ kx = workLimiterBandTable[loLimIndex] + lowSubband;
+
+ temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
+ temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);
+
+ if (temp < FL2FXCONST_DBL (0.49f)>>5) {
+ if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
+ workLimiterBandTable[hiLimIndex] = highSubband;
+ nBands--;
+ hiLimIndex++;
+ continue;
+ }
+ isPatchBorder[0] = isPatchBorder[1] = 0;
+ for (k = 0; k <= noPatches; k++) {
+ if (workLimiterBandTable[hiLimIndex] == patchBorders[k]) {
+ isPatchBorder[1] = 1;
+ break;
+ }
+ }
+ if (!isPatchBorder[1]) {
+ workLimiterBandTable[hiLimIndex] = highSubband;
+ nBands--;
+ hiLimIndex++;
+ continue;
+ }
+ for (k = 0; k <= noPatches; k++) {
+ if (workLimiterBandTable[loLimIndex] == patchBorders[k]) {
+ isPatchBorder[0] = 1;
+ break;
+ }
+ }
+ if (!isPatchBorder[0]) {
+ workLimiterBandTable[loLimIndex] = highSubband;
+ nBands--;
+ }
+ }
+ loLimIndex = hiLimIndex;
+ hiLimIndex++;
+
+ }
+ shellsort(workLimiterBandTable, tempNoLim + 1);
+
+ /* Test if algorithm exceeded maximum allowed limiterbands */
+ if( nBands > MAX_NUM_LIMITERS || nBands <= 0) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ /* Copy limiterbands from working buffer into final destination */
+ for (k = 0; k <= nBands; k++) {
+ limiterBandTable[k] = workLimiterBandTable[k];
+ }
+ }
+ *noLimiterBands = nBands;
+
+ return SBRDEC_OK;
+}
+
diff --git a/libSBRdec/src/env_calc.h b/libSBRdec/src/env_calc.h
new file mode 100644
index 0000000..d21e0b3
--- /dev/null
+++ b/libSBRdec/src/env_calc.h
@@ -0,0 +1,165 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Envelope calculation prototypes
+*/
+#ifndef __ENV_CALC_H
+#define __ENV_CALC_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h" /* for HANDLE_SBR_HEADER_DATA */
+#include "sbr_scale.h"
+
+
+typedef struct
+{
+ FIXP_DBL filtBuffer[MAX_FREQ_COEFFS]; /*!< previous gains (required for smoothing) */
+ FIXP_DBL filtBufferNoise[MAX_FREQ_COEFFS]; /*!< previous noise levels (required for smoothing) */
+ SCHAR filtBuffer_e[MAX_FREQ_COEFFS]; /*!< Exponents of previous gains */
+ SCHAR filtBufferNoise_e; /*!< Common exponent of previous noise levels */
+
+ int startUp; /*!< flag to signal initial conditions in buffers */
+ int phaseIndex; /*!< Index for randomPase array */
+ int prevTranEnv; /*!< The transient envelope of the previous frame. */
+
+ int harmFlagsPrev[(MAX_FREQ_COEFFS+15)/16];
+ /*!< Words with 16 flags each indicating where a sine was added in the previous frame.*/
+ UCHAR harmIndex; /*!< Current phase of synthetic sine */
+
+}
+SBR_CALCULATE_ENVELOPE;
+
+typedef SBR_CALCULATE_ENVELOPE *HANDLE_SBR_CALCULATE_ENVELOPE;
+
+
+
+void
+calculateSbrEnvelope (QMF_SCALE_FACTOR *sbrScaleFactor,
+ HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameData,
+ FIXP_DBL **analysBufferReal,
+ FIXP_DBL **analysBufferImag, /*!< Imag part of subband samples to be processed */
+ const int useLP,
+ FIXP_DBL *degreeAlias, /*!< Estimated aliasing for each QMF channel */
+ const UINT flags,
+ const int frameErrorFlag
+ );
+
+SBR_ERROR
+createSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ const int chan,
+ const UINT flags);
+
+int
+deleteSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hSbrCalculateEnvelope);
+
+void
+resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv);
+
+SBR_ERROR
+ResetLimiterBands ( UCHAR *limiterBandTable,
+ UCHAR *noLimiterBands,
+ UCHAR *freqBandTable,
+ int noFreqBands,
+ const PATCH_PARAM *patchParam,
+ int noPatches,
+ int limiterBands);
+
+void rescaleSubbandSamples( FIXP_DBL ** re,
+ FIXP_DBL ** im,
+ int lowSubband, int noSubbands,
+ int start_pos, int next_pos,
+ int shift);
+
+FIXP_DBL maxSubbandSample( FIXP_DBL ** analysBufferReal_m,
+ FIXP_DBL ** analysBufferImag_m,
+ int lowSubband,
+ int highSubband,
+ int start_pos,
+ int stop_pos);
+
+#endif // __ENV_CALC_H
diff --git a/libSBRdec/src/env_dec.cpp b/libSBRdec/src/env_dec.cpp
new file mode 100644
index 0000000..ac6c299
--- /dev/null
+++ b/libSBRdec/src/env_dec.cpp
@@ -0,0 +1,852 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief envelope decoding
+ This module provides envelope decoding and error concealment algorithms. The main
+ entry point is decodeSbrData().
+
+ \sa decodeSbrData(),\ref documentationOverview
+*/
+
+#include "env_dec.h"
+
+#include "env_extr.h"
+#include "transcendent.h"
+
+#include "genericStds.h"
+
+
+static void decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_sbr_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_otherChannel);
+static void sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_data_left,
+ HANDLE_SBR_FRAME_DATA h_data_right);
+static void requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data,
+ int ampResolution);
+static void deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_sbr_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static void decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_sbr_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static void timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_sbr_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+static int checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_sbr_data,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data);
+
+
+
+#define SBR_ENERGY_PAN_OFFSET (12 << ENV_EXP_FRACT)
+#define SBR_MAX_ENERGY (35 << ENV_EXP_FRACT)
+
+#define DECAY ( 1 << ENV_EXP_FRACT)
+
+#if ENV_EXP_FRACT
+#define DECAY_COUPLING ( 1 << (ENV_EXP_FRACT-1) ) /*!< corresponds to a value of 0.5 */
+#else
+#define DECAY_COUPLING 1 /*!< If the energy data is not shifted, use 1 instead of 0.5 */
+#endif
+
+
+/*!
+ \brief Convert table index
+*/
+static int indexLow2High(int offset, /*!< mapping factor */
+ int index, /*!< index to scalefactor band */
+ int res) /*!< frequency resolution */
+{
+ if(res == 0)
+ {
+ if (offset >= 0)
+ {
+ if (index < offset)
+ return(index);
+ else
+ return(2*index - offset);
+ }
+ else
+ {
+ offset = -offset;
+ if (index < offset)
+ return(2*index+index);
+ else
+ return(2*index + offset);
+ }
+ }
+ else
+ return(index);
+}
+
+
+/*!
+ \brief Update previous envelope value for delta-coding
+
+ The current envelope values needs to be stored for delta-coding
+ in the next frame. The stored envelope is always represented with
+ the high frequency resolution. If the current envelope uses the
+ low frequency resolution, the energy value will be mapped to the
+ corresponding high-res bands.
+*/
+static void mapLowResEnergyVal(FIXP_SGL currVal, /*!< current energy value */
+ FIXP_SGL* prevData,/*!< pointer to previous data vector */
+ int offset, /*!< mapping factor */
+ int index, /*!< index to scalefactor band */
+ int res) /*!< frequeny resolution */
+{
+ if(res == 0)
+ {
+ if (offset >= 0)
+ {
+ if(index < offset)
+ prevData[index] = currVal;
+ else
+ {
+ prevData[2*index - offset] = currVal;
+ prevData[2*index+1 - offset] = currVal;
+ }
+ }
+ else
+ {
+ offset = -offset;
+ if (index < offset)
+ {
+ prevData[3*index] = currVal;
+ prevData[3*index+1] = currVal;
+ prevData[3*index+2] = currVal;
+ }
+ else
+ {
+ prevData[2*index + offset] = currVal;
+ prevData[2*index + 1 + offset] = currVal;
+ }
+ }
+ }
+ else
+ prevData[index] = currVal;
+}
+
+
+
+/*!
+ \brief Convert raw envelope and noisefloor data to energy levels
+
+ This function is being called by sbrDecoder_ParseElement() and provides two important algorithms:
+
+ First the function decodes envelopes and noise floor levels as described in requantizeEnvelopeData()
+ and sbr_envelope_unmapping(). The function also implements concealment algorithms in case there are errors
+ within the sbr data. For both operations fractional arithmetic is used.
+ Therefore you might encounter different output values on your target
+ system compared to the reference implementation.
+*/
+void
+decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel frame data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left, /*!< pointer to left channel previous frame data */
+ HANDLE_SBR_FRAME_DATA h_data_right, /*!< pointer to right channel frame data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right)/*!< pointer to right channel previous frame data */
+{
+ FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS];
+ int errLeft;
+
+ /* Save previous energy values to be able to reuse them later for concealment. */
+ FDKmemcpy (tempSfbNrgPrev, h_prev_data_left->sfb_nrg_prev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL));
+
+ decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right);
+ decodeNoiseFloorlevels (hHeaderData, h_data_left, h_prev_data_left);
+
+ if(h_data_right != NULL) {
+ errLeft = hHeaderData->frameErrorFlag;
+ decodeEnvelope (hHeaderData, h_data_right, h_prev_data_right, h_prev_data_left);
+ decodeNoiseFloorlevels (hHeaderData, h_data_right, h_prev_data_right);
+
+ if (!errLeft && hHeaderData->frameErrorFlag) {
+ /* If an error occurs in the right channel where the left channel seemed ok,
+ we apply concealment also on the left channel. This ensures that the coupling
+ modes of both channels match and that we have the same number of envelopes in
+ coupling mode.
+ However, as the left channel has already been processed before, the resulting
+ energy levels are not the same as if the left channel had been concealed
+ during the first call of decodeEnvelope().
+ */
+ /* Restore previous energy values for concealment, because the values have been
+ overwritten by the first call of decodeEnvelope(). */
+ FDKmemcpy (h_prev_data_left->sfb_nrg_prev, tempSfbNrgPrev, MAX_FREQ_COEFFS * sizeof(FIXP_SGL));
+ /* Do concealment */
+ decodeEnvelope (hHeaderData, h_data_left, h_prev_data_left, h_prev_data_right);
+ }
+
+ if (h_data_left->coupling) {
+ sbr_envelope_unmapping (hHeaderData, h_data_left, h_data_right);
+ }
+ }
+
+ /* Display the data for debugging: */
+}
+
+
+/*!
+ \brief Convert from coupled channels to independent L/R data
+*/
+static void
+sbr_envelope_unmapping (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_data_left, /*!< pointer to left channel */
+ HANDLE_SBR_FRAME_DATA h_data_right) /*!< pointer to right channel */
+{
+ int i;
+ FIXP_SGL tempL_m, tempR_m, tempRplus1_m, newL_m, newR_m;
+ SCHAR tempL_e, tempR_e, tempRplus1_e, newL_e, newR_e;
+
+
+ /* 1. Unmap (already dequantized) coupled envelope energies */
+
+ for (i = 0; i < h_data_left->nScaleFactors; i++) {
+ tempR_m = (FIXP_SGL)((LONG)h_data_right->iEnvelope[i] & MASK_M);
+ tempR_e = (SCHAR)((LONG)h_data_right->iEnvelope[i] & MASK_E);
+
+ tempR_e -= (18 + NRG_EXP_OFFSET); /* -18 = ld(UNMAPPING_SCALE / h_data_right->nChannels) */
+ tempL_m = (FIXP_SGL)((LONG)h_data_left->iEnvelope[i] & MASK_M);
+ tempL_e = (SCHAR)((LONG)h_data_left->iEnvelope[i] & MASK_E);
+
+ tempL_e -= NRG_EXP_OFFSET;
+
+ /* Calculate tempRight+1 */
+ FDK_add_MantExp( tempR_m, tempR_e,
+ FL2FXCONST_SGL(0.5f), 1, /* 1.0 */
+ &tempRplus1_m, &tempRplus1_e);
+
+ FDK_divide_MantExp( tempL_m, tempL_e+1, /* 2 * tempLeft */
+ tempRplus1_m, tempRplus1_e,
+ &newR_m, &newR_e );
+
+ if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) {
+ newR_m >>= 1;
+ newR_e += 1;
+ }
+
+ newL_m = FX_DBL2FX_SGL(fMult(tempR_m,newR_m));
+ newL_e = tempR_e + newR_e;
+
+ h_data_right->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) +
+ (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NRG_EXP_OFFSET) & MASK_E);
+ h_data_left->iEnvelope[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) +
+ (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NRG_EXP_OFFSET) & MASK_E);
+ }
+
+ /* 2. Dequantize and unmap coupled noise floor levels */
+
+ for (i = 0; i < hHeaderData->freqBandData.nNfb * h_data_left->frameInfo.nNoiseEnvelopes; i++) {
+
+ tempL_e = (SCHAR)(6 - (LONG)h_data_left->sbrNoiseFloorLevel[i]);
+ tempR_e = (SCHAR)((LONG)h_data_right->sbrNoiseFloorLevel[i] - 12) /*SBR_ENERGY_PAN_OFFSET*/;
+
+ /* Calculate tempR+1 */
+ FDK_add_MantExp( FL2FXCONST_SGL(0.5f), 1+tempR_e, /* tempR */
+ FL2FXCONST_SGL(0.5f), 1, /* 1.0 */
+ &tempRplus1_m, &tempRplus1_e);
+
+ /* Calculate 2*tempLeft/(tempR+1) */
+ FDK_divide_MantExp( FL2FXCONST_SGL(0.5f), tempL_e+2, /* 2 * tempLeft */
+ tempRplus1_m, tempRplus1_e,
+ &newR_m, &newR_e );
+
+ /* if (newR_m >= ((FIXP_SGL)MAXVAL_SGL - ROUNDING)) {
+ newR_m >>= 1;
+ newR_e += 1;
+ } */
+
+ /* L = tempR * R */
+ newL_m = newR_m;
+ newL_e = newR_e + tempR_e;
+ h_data_right->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newR_m + ROUNDING) & MASK_M)) +
+ (FIXP_SGL)((SHORT)(FIXP_SGL)(newR_e + NOISE_EXP_OFFSET) & MASK_E);
+ h_data_left->sbrNoiseFloorLevel[i] = ((FIXP_SGL)((SHORT)(FIXP_SGL)(newL_m + ROUNDING) & MASK_M)) +
+ (FIXP_SGL)((SHORT)(FIXP_SGL)(newL_e + NOISE_EXP_OFFSET) & MASK_E);
+ }
+}
+
+
+/*!
+ \brief Simple alternative to the real SBR concealment
+
+ If the real frameInfo is not available due to a frame loss, a replacement will
+ be constructed with 1 envelope spanning the whole frame (FIX-FIX).
+ The delta-coded energies are set to negative values, resulting in a fade-down.
+ In case of coupling, the balance-channel will move towards the center.
+*/
+static void
+leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */
+ )
+{
+ FIXP_SGL target; /* targeted level for sfb_nrg_prev during fade-down */
+ FIXP_SGL step; /* speed of fade */
+ int i;
+
+ int currentStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+ int currentStopPos = hHeaderData->numberTimeSlots;
+
+
+ /* Use some settings of the previous frame */
+ h_sbr_data->ampResolutionCurrentFrame = h_prev_data->ampRes;
+ h_sbr_data->coupling = h_prev_data->coupling;
+ for(i=0;i<MAX_INVF_BANDS;i++)
+ h_sbr_data->sbr_invf_mode[i] = h_prev_data->sbr_invf_mode[i];
+
+ /* Generate concealing control data */
+
+ h_sbr_data->frameInfo.nEnvelopes = 1;
+ h_sbr_data->frameInfo.borders[0] = currentStartPos;
+ h_sbr_data->frameInfo.borders[1] = currentStopPos;
+ h_sbr_data->frameInfo.freqRes[0] = 1;
+ h_sbr_data->frameInfo.tranEnv = -1; /* no transient */
+ h_sbr_data->frameInfo.nNoiseEnvelopes = 1;
+ h_sbr_data->frameInfo.bordersNoise[0] = currentStartPos;
+ h_sbr_data->frameInfo.bordersNoise[1] = currentStopPos;
+
+ h_sbr_data->nScaleFactors = hHeaderData->freqBandData.nSfb[1];
+
+ /* Generate fake envelope data */
+
+ h_sbr_data->domain_vec[0] = 1;
+
+ if (h_sbr_data->coupling == COUPLING_BAL) {
+ target = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET;
+ step = (FIXP_SGL)DECAY_COUPLING;
+ }
+ else {
+ target = FL2FXCONST_SGL(0.0f);
+ step = (FIXP_SGL)DECAY;
+ }
+ if (hHeaderData->bs_info.ampResolution == 0) {
+ target <<= 1;
+ step <<= 1;
+ }
+
+ for (i=0; i < h_sbr_data->nScaleFactors; i++) {
+ if (h_prev_data->sfb_nrg_prev[i] > target)
+ h_sbr_data->iEnvelope[i] = -step;
+ else
+ h_sbr_data->iEnvelope[i] = step;
+ }
+
+ /* Noisefloor levels are always cleared ... */
+
+ h_sbr_data->domain_vec_noise[0] = 1;
+ for (i=0; i < hHeaderData->freqBandData.nNfb; i++)
+ h_sbr_data->sbrNoiseFloorLevel[i] = FL2FXCONST_SGL(0.0f);
+
+ /* ... and so are the sines */
+ FDKmemclear(h_sbr_data->addHarmonics, MAX_FREQ_COEFFS);
+}
+
+
+/*!
+ \brief Build reference energies and noise levels from bitstream elements
+*/
+static void
+decodeEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< pointer to data of last frame */
+ HANDLE_SBR_PREV_FRAME_DATA otherChannel /*!< other channel's last frame data */
+ )
+{
+ int i;
+ int fFrameError = hHeaderData->frameErrorFlag;
+ FIXP_SGL tempSfbNrgPrev[MAX_FREQ_COEFFS];
+
+ if (!fFrameError) {
+ /*
+ To avoid distortions after bad frames, set the error flag if delta coding in time occurs.
+ However, SBR can take a little longer to come up again.
+ */
+ if ( h_prev_data->frameErrorFlag ) {
+ if (h_sbr_data->domain_vec[0] != 0) {
+ fFrameError = 1;
+ }
+ } else {
+ /* Check that the previous stop position and the current start position match.
+ (Could be done in checkFrameInfo(), but the previous frame data is not available there) */
+ if ( h_sbr_data->frameInfo.borders[0] != h_prev_data->stopPos - hHeaderData->numberTimeSlots ) {
+ /* Both the previous as well as the current frame are flagged to be ok, but they do not match! */
+ if (h_sbr_data->domain_vec[0] == 1) {
+ /* Prefer concealment over delta-time coding between the mismatching frames */
+ fFrameError = 1;
+ }
+ else {
+ /* Close the gap in time by triggering timeCompensateFirstEnvelope() */
+ fFrameError = 1;
+ }
+ }
+ }
+ }
+
+
+ if (fFrameError) /* Error is detected */
+ {
+ leanSbrConcealment(hHeaderData,
+ h_sbr_data,
+ h_prev_data);
+
+ /* decode the envelope data to linear PCM */
+ deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data);
+ }
+ else /*Do a temporary dummy decoding and check that the envelope values are within limits */
+ {
+ if (h_prev_data->frameErrorFlag) {
+ timeCompensateFirstEnvelope (hHeaderData, h_sbr_data, h_prev_data);
+ if (h_sbr_data->coupling != h_prev_data->coupling) {
+ /*
+ Coupling mode has changed during concealment.
+ The stored energy levels need to be converted.
+ */
+ for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) {
+ /* Former Level-Channel will be used for both channels */
+ if (h_prev_data->coupling == COUPLING_BAL)
+ h_prev_data->sfb_nrg_prev[i] = otherChannel->sfb_nrg_prev[i];
+ /* Former L/R will be combined as the new Level-Channel */
+ else if (h_sbr_data->coupling == COUPLING_LEVEL)
+ h_prev_data->sfb_nrg_prev[i] = (h_prev_data->sfb_nrg_prev[i] + otherChannel->sfb_nrg_prev[i]) >> 1;
+ else if (h_sbr_data->coupling == COUPLING_BAL)
+ h_prev_data->sfb_nrg_prev[i] = (FIXP_SGL)SBR_ENERGY_PAN_OFFSET;
+ }
+ }
+ }
+ FDKmemcpy (tempSfbNrgPrev, h_prev_data->sfb_nrg_prev,
+ MAX_FREQ_COEFFS * sizeof (FIXP_SGL));
+
+ deltaToLinearPcmEnvelopeDecoding (hHeaderData, h_sbr_data, h_prev_data);
+
+ fFrameError = checkEnvelopeData (hHeaderData, h_sbr_data, h_prev_data);
+
+ if (fFrameError)
+ {
+ hHeaderData->frameErrorFlag = 1;
+ FDKmemcpy (h_prev_data->sfb_nrg_prev, tempSfbNrgPrev,
+ MAX_FREQ_COEFFS * sizeof (FIXP_SGL));
+ decodeEnvelope (hHeaderData, h_sbr_data, h_prev_data, otherChannel);
+ return;
+ }
+ }
+
+ requantizeEnvelopeData (h_sbr_data, h_sbr_data->ampResolutionCurrentFrame);
+
+ hHeaderData->frameErrorFlag = fFrameError;
+}
+
+
+/*!
+ \brief Verify that envelope energies are within the allowed range
+ \return 0 if all is fine, 1 if an envelope value was too high
+*/
+static int
+checkEnvelopeData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data /*!< pointer to data of last frame */
+ )
+{
+ FIXP_SGL *iEnvelope = h_sbr_data->iEnvelope;
+ FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev;
+ int i = 0, errorFlag = 0;
+ FIXP_SGL sbr_max_energy =
+ (h_sbr_data->ampResolutionCurrentFrame == 1) ? SBR_MAX_ENERGY : (SBR_MAX_ENERGY << 1);
+
+ /*
+ Range check for current energies
+ */
+ for (i = 0; i < h_sbr_data->nScaleFactors; i++) {
+ if (iEnvelope[i] > sbr_max_energy) {
+ errorFlag = 1;
+ }
+ if (iEnvelope[i] < FL2FXCONST_SGL(0.0f)) {
+ errorFlag = 1;
+ /* iEnvelope[i] = FL2FXCONST_SGL(0.0f); */
+ }
+ }
+
+ /*
+ Range check for previous energies
+ */
+ for (i = 0; i < hHeaderData->freqBandData.nSfb[1]; i++) {
+ sfb_nrg_prev[i] = fixMax(sfb_nrg_prev[i], FL2FXCONST_SGL(0.0f));
+ sfb_nrg_prev[i] = fixMin(sfb_nrg_prev[i], sbr_max_energy);
+ }
+
+ return (errorFlag);
+}
+
+
+/*!
+ \brief Verify that the noise levels are within the allowed range
+
+ The function is equivalent to checkEnvelopeData().
+ When the noise-levels are being decoded, it is already too late for
+ concealment. Therefore the noise levels are simply limited here.
+*/
+static void
+limitNoiseLevels(HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data) /*!< pointer to current data */
+{
+ int i;
+ int nNfb = hHeaderData->freqBandData.nNfb;
+
+ /*
+ Set range limits. The exact values depend on the coupling mode.
+ However this limitation is primarily intended to avoid unlimited
+ accumulation of the delta-coded noise levels.
+ */
+ #define lowerLimit ((FIXP_SGL)0) /* lowerLimit actually refers to the _highest_ noise energy */
+ #define upperLimit ((FIXP_SGL)35) /* upperLimit actually refers to the _lowest_ noise energy */
+
+ /*
+ Range check for current noise levels
+ */
+ for (i = 0; i < h_sbr_data->frameInfo.nNoiseEnvelopes * nNfb; i++) {
+ h_sbr_data->sbrNoiseFloorLevel[i] = fixMin(h_sbr_data->sbrNoiseFloorLevel[i], upperLimit);
+ h_sbr_data->sbrNoiseFloorLevel[i] = fixMax(h_sbr_data->sbrNoiseFloorLevel[i], lowerLimit);
+ }
+}
+
+
+/*!
+ \brief Compensate for the wrong timing that might occur after a frame error.
+*/
+static void
+timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to actual data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to data of last frame */
+{
+ int i, nScalefactors;
+ FRAME_INFO *pFrameInfo = &h_sbr_data->frameInfo;
+ UCHAR *nSfb = hHeaderData->freqBandData.nSfb;
+ int estimatedStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+ int refLen, newLen, shift;
+ FIXP_SGL deltaExp;
+
+ /* Original length of first envelope according to bitstream */
+ refLen = pFrameInfo->borders[1] - pFrameInfo->borders[0];
+ /* Corrected length of first envelope (concealing can make the first envelope longer) */
+ newLen = pFrameInfo->borders[1] - estimatedStartPos;
+
+ if (newLen <= 0) {
+ /* An envelope length of <= 0 would not work, so we don't use it.
+ May occur if the previous frame was flagged bad due to a mismatch
+ of the old and new frame infos. */
+ newLen = refLen;
+ estimatedStartPos = pFrameInfo->borders[0];
+ }
+
+ deltaExp = FDK_getNumOctavesDiv8(newLen, refLen);
+
+ /* Shift by -3 to rescale ld-table, 1-ampRes to enable coarser steps */
+ shift = (FRACT_BITS - 1 - ENV_EXP_FRACT + 1 - h_sbr_data->ampResolutionCurrentFrame - 3);
+ deltaExp = deltaExp >> shift;
+ pFrameInfo->borders[0] = estimatedStartPos;
+ pFrameInfo->bordersNoise[0] = estimatedStartPos;
+
+ if (h_sbr_data->coupling != COUPLING_BAL) {
+ nScalefactors = (pFrameInfo->freqRes[0]) ? nSfb[1] : nSfb[0];
+
+ for (i = 0; i < nScalefactors; i++)
+ h_sbr_data->iEnvelope[i] = h_sbr_data->iEnvelope[i] + deltaExp;
+ }
+}
+
+
+
+/*!
+ \brief Convert each envelope value from logarithmic to linear domain
+
+ Energy levels are transmitted in powers of 2, i.e. only the exponent
+ is extracted from the bitstream.
+ Therefore, normally only integer exponents can occur. However during
+ fading (in case of a corrupt bitstream), a fractional part can also
+ occur. The data in the array iEnvelope is shifted left by ENV_EXP_FRACT
+ compared to an integer representation so that numbers smaller than 1
+ can be represented.
+
+ This function calculates a mantissa corresponding to the fractional
+ part of the exponent for each reference energy. The array iEnvelope
+ is converted in place to save memory. Input and output data must
+ be interpreted differently, as shown in the below figure:
+
+ \image html EnvelopeData.png
+
+ The data is then used in calculateSbrEnvelope().
+*/
+static void
+requantizeEnvelopeData (HANDLE_SBR_FRAME_DATA h_sbr_data, int ampResolution)
+{
+ int i;
+ FIXP_SGL mantissa;
+ int ampShift = 1 - ampResolution;
+ int exponent;
+
+ /* In case that ENV_EXP_FRACT is changed to something else but 0 or 8,
+ the initialization of this array has to be adapted!
+ */
+#if ENV_EXP_FRACT
+ static const FIXP_SGL pow2[ENV_EXP_FRACT] =
+ {
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 1))), /* 0.7071 */
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 2))), /* 0.5946 */
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 3))),
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 4))),
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 5))),
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 6))),
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 7))),
+ FL2FXCONST_SGL(0.5f * pow(2.0f, pow(0.5f, 8))) /* 0.5013 */
+ };
+
+ int bit, mask;
+#endif
+
+ for (i = 0; i < h_sbr_data->nScaleFactors; i++) {
+ exponent = (LONG)h_sbr_data->iEnvelope[i];
+
+#if ENV_EXP_FRACT
+
+ exponent = exponent >> ampShift;
+ mantissa = 0.5f;
+
+ /* Amplify mantissa according to the fractional part of the
+ exponent (result will be between 0.500000 and 0.999999)
+ */
+ mask = 1; /* begin with lowest bit of exponent */
+
+ for ( bit=ENV_EXP_FRACT-1; bit>=0; bit-- ) {
+ if (exponent & mask) {
+ /* The current bit of the exponent is set,
+ multiply mantissa with the corresponding factor: */
+ mantissa = (FIXP_SGL)( (mantissa * pow2[bit]) << 1);
+ }
+ /* Advance to next bit */
+ mask = mask << 1;
+ }
+
+ /* Make integer part of exponent right aligned */
+ exponent = exponent >> ENV_EXP_FRACT;
+
+#else
+ /* In case of the high amplitude resolution, 1 bit of the exponent gets lost by the shift.
+ This will be compensated by a mantissa of 0.5*sqrt(2) instead of 0.5 if that bit is 1. */
+ mantissa = (exponent & ampShift) ? FL2FXCONST_SGL(0.707106781186548f) : FL2FXCONST_SGL(0.5f);
+ exponent = exponent >> ampShift;
+#endif
+
+ /*
+ Mantissa was set to 0.5 (instead of 1.0, therefore increase exponent by 1).
+ Multiply by L=nChannels=64 by increasing exponent by another 6.
+ => Increase exponent by 7
+ */
+ exponent += 7 + NRG_EXP_OFFSET;
+
+ /* Combine mantissa and exponent and write back the result */
+ h_sbr_data->iEnvelope[i] = (FIXP_SGL)(((LONG)mantissa & MASK_M) | (exponent & MASK_E));
+
+ }
+}
+
+
+/*!
+ \brief Build new reference energies from old ones and delta coded data
+*/
+static void
+deltaToLinearPcmEnvelopeDecoding (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */
+{
+ int i, domain, no_of_bands, band, freqRes;
+
+ FIXP_SGL *sfb_nrg_prev = h_prev_data->sfb_nrg_prev;
+ FIXP_SGL *ptr_nrg = h_sbr_data->iEnvelope;
+
+ int offset = 2 * hHeaderData->freqBandData.nSfb[0] - hHeaderData->freqBandData.nSfb[1];
+
+ for (i = 0; i < h_sbr_data->frameInfo.nEnvelopes; i++) {
+ domain = h_sbr_data->domain_vec[i];
+ freqRes = h_sbr_data->frameInfo.freqRes[i];
+
+ FDK_ASSERT(freqRes >= 0 && freqRes <= 1);
+
+ no_of_bands = hHeaderData->freqBandData.nSfb[freqRes];
+
+ FDK_ASSERT(no_of_bands < (64));
+
+ if (domain == 0)
+ {
+ mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, 0, freqRes);
+ ptr_nrg++;
+ for (band = 1; band < no_of_bands; band++)
+ {
+ *ptr_nrg = *ptr_nrg + *(ptr_nrg-1);
+ mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes);
+ ptr_nrg++;
+ }
+ }
+ else
+ {
+ for (band = 0; band < no_of_bands; band++)
+ {
+ *ptr_nrg = *ptr_nrg + sfb_nrg_prev[indexLow2High(offset, band, freqRes)];
+ mapLowResEnergyVal(*ptr_nrg, sfb_nrg_prev, offset, band, freqRes);
+ ptr_nrg++;
+ }
+ }
+ }
+}
+
+
+/*!
+ \brief Build new noise levels from old ones and delta coded data
+*/
+static void
+decodeNoiseFloorlevels (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_sbr_data, /*!< pointer to current data */
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data) /*!< pointer to previous data */
+{
+ int i;
+ int nNfb = hHeaderData->freqBandData.nNfb;
+ int nNoiseFloorEnvelopes = h_sbr_data->frameInfo.nNoiseEnvelopes;
+
+ /* Decode first noise envelope */
+
+ if (h_sbr_data->domain_vec_noise[0] == 0) {
+ FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[0];
+ for (i = 1; i < nNfb; i++) {
+ noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i];
+ h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel;
+ }
+ }
+ else {
+ for (i = 0; i < nNfb; i++) {
+ h_sbr_data->sbrNoiseFloorLevel[i] += h_prev_data->prevNoiseLevel[i];
+ }
+ }
+
+ /* If present, decode the second noise envelope
+ Note: nNoiseFloorEnvelopes can only be 1 or 2 */
+
+ if (nNoiseFloorEnvelopes > 1) {
+ if (h_sbr_data->domain_vec_noise[1] == 0) {
+ FIXP_SGL noiseLevel = h_sbr_data->sbrNoiseFloorLevel[nNfb];
+ for (i = nNfb + 1; i < 2*nNfb; i++) {
+ noiseLevel += h_sbr_data->sbrNoiseFloorLevel[i];
+ h_sbr_data->sbrNoiseFloorLevel[i] = noiseLevel;
+ }
+ }
+ else {
+ for (i = 0; i < nNfb; i++) {
+ h_sbr_data->sbrNoiseFloorLevel[i + nNfb] += h_sbr_data->sbrNoiseFloorLevel[i];
+ }
+ }
+ }
+
+ limitNoiseLevels(hHeaderData, h_sbr_data);
+
+ /* Update prevNoiseLevel with the last noise envelope */
+ for (i = 0; i < nNfb; i++)
+ h_prev_data->prevNoiseLevel[i] = h_sbr_data->sbrNoiseFloorLevel[i + nNfb*(nNoiseFloorEnvelopes-1)];
+
+
+ /* Requantize the noise floor levels in COUPLING_OFF-mode */
+ if (!h_sbr_data->coupling) {
+ int nf_e;
+
+ for (i = 0; i < nNoiseFloorEnvelopes*nNfb; i++) {
+ nf_e = 6 - (LONG)h_sbr_data->sbrNoiseFloorLevel[i] + 1 + NOISE_EXP_OFFSET;
+ /* +1 to compensate for a mantissa of 0.5 instead of 1.0 */
+
+ h_sbr_data->sbrNoiseFloorLevel[i] =
+ (FIXP_SGL)( ((LONG)FL2FXCONST_SGL(0.5f)) + /* mantissa */
+ (nf_e & MASK_E) ); /* exponent */
+
+ }
+ }
+}
diff --git a/libSBRdec/src/env_dec.h b/libSBRdec/src/env_dec.h
new file mode 100644
index 0000000..3e656ed
--- /dev/null
+++ b/libSBRdec/src/env_dec.h
@@ -0,0 +1,101 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Envelope decoding
+*/
+#ifndef __ENV_DEC_H
+#define __ENV_DEC_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+
+void decodeSbrData (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_data_left,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_left,
+ HANDLE_SBR_FRAME_DATA h_data_right,
+ HANDLE_SBR_PREV_FRAME_DATA h_prev_data_right);
+
+
+#endif
diff --git a/libSBRdec/src/env_extr.cpp b/libSBRdec/src/env_extr.cpp
new file mode 100644
index 0000000..716fb91
--- /dev/null
+++ b/libSBRdec/src/env_extr.cpp
@@ -0,0 +1,1395 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Envelope extraction
+ The functions provided by this module are mostly called by applySBR(). After it is
+ determined that there is valid SBR data, sbrGetHeaderData() might be called if the current
+ SBR data contains an \ref SBR_HEADER_ELEMENT as opposed to a \ref SBR_STANDARD_ELEMENT. This function
+ may return various error codes as defined in #SBR_HEADER_STATUS . Most importantly it returns HEADER_RESET when decoder
+ settings need to be recalculated according to the SBR specifications. In that case applySBR()
+ will initiatite the required re-configuration.
+
+ The header data is stored in a #SBR_HEADER_DATA structure.
+
+ The actual SBR data for the current frame is decoded into SBR_FRAME_DATA stuctures by sbrGetChannelPairElement()
+ [for stereo streams] and sbrGetSingleChannelElement() [for mono streams]. There is no fractional arithmetic involved.
+
+ Once the information is extracted, the data needs to be further prepared before the actual decoding process.
+ This is done in decodeSbrData().
+
+ \sa Description of buffer management in applySBR(). \ref documentationOverview
+
+ <h1>About the SBR data format:</h1>
+
+ Each frame includes SBR data (side chain information), and can be either the \ref SBR_HEADER_ELEMENT or the \ref SBR_STANDARD_ELEMENT.
+ Parts of the data can be protected by a CRC checksum.
+
+ \anchor SBR_HEADER_ELEMENT <h2>The SBR_HEADER_ELEMENT</h2>
+
+ The SBR_HEADER_ELEMENT can be transmitted with every frame, however, it typically is send every second or so. It contains fundamental
+ information such as SBR sampling frequency and frequency range as well as control signals that do not require frequent changes. It also
+ includes the \ref SBR_STANDARD_ELEMENT.
+
+ Depending on the changes between the information in a current SBR_HEADER_ELEMENT and the previous SBR_HEADER_ELEMENT, the SBR decoder might need
+ to be reset and reconfigured (e.g. new tables need to be calculated).
+
+ \anchor SBR_STANDARD_ELEMENT <h2>The SBR_STANDARD_ELEMENT</h2>
+
+ This data can be subdivided into "side info" and "raw data", where side info is defined as signals needed to decode the raw data
+ and some decoder tuning signals. Raw data is referred to as PCM and Huffman coded envelope and noise floor estimates. The side info also
+ includes information about the time-frequency grid for the current frame.
+
+ \sa \ref documentationOverview
+*/
+
+#include "env_extr.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+#include "huff_dec.h"
+
+
+#include "psbitdec.h"
+
+#define DRM_PARAMETRIC_STEREO 0
+#define EXTENSION_ID_PS_CODING 2
+
+
+static int extractFrameInfo (HANDLE_FDK_BITSTREAM hBs,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_frame_data,
+ const UINT nrOfChannels,
+ const UINT flags
+ );
+
+
+static int sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_frame_data,
+ HANDLE_FDK_BITSTREAM hBs,
+ const UINT flags);
+
+static void sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA hFrameData,
+ HANDLE_FDK_BITSTREAM hBs);
+
+static void sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_frame_data,
+ HANDLE_FDK_BITSTREAM hBs);
+
+static int checkFrameInfo (FRAME_INFO *pFrameInfo, int numberOfTimeSlots, int overlap, int timeStep);
+
+SBR_ERROR
+initHeaderData (
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ const int sampleRateIn,
+ const int sampleRateOut,
+ const int samplesPerFrame,
+ const UINT flags
+ )
+{
+ HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+ SBR_ERROR sbrError = SBRDEC_OK;
+ int numAnalysisBands;
+
+ if ( sampleRateIn == sampleRateOut ) {
+ hHeaderData->sbrProcSmplRate = sampleRateOut<<1;
+ numAnalysisBands = 32;
+ } else {
+ hHeaderData->sbrProcSmplRate = sampleRateOut;
+ if ( (sampleRateOut>>1) == sampleRateIn) {
+ /* 1:2 */
+ numAnalysisBands = 32;
+ } else if ( (sampleRateOut>>2) == sampleRateIn ) {
+ /* 1:4 */
+ numAnalysisBands = 32;
+ } else if ( (sampleRateOut*3)>>3 == (sampleRateIn*8)>>3 ) {
+ /* 3:8, 3/4 core frame length */
+ numAnalysisBands = 24;
+ } else {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+ }
+
+ /* Fill in default values first */
+ hHeaderData->syncState = SBR_NOT_INITIALIZED;
+ hHeaderData->status = 0;
+ hHeaderData->frameErrorFlag = 0;
+
+ hHeaderData->bs_info.ampResolution = 1;
+ hHeaderData->bs_info.xover_band = 0;
+ hHeaderData->bs_info.sbr_preprocessing = 0;
+
+ hHeaderData->bs_data.startFreq = 5;
+ hHeaderData->bs_data.stopFreq = 0;
+ hHeaderData->bs_data.freqScale = 2;
+ hHeaderData->bs_data.alterScale = 1;
+ hHeaderData->bs_data.noise_bands = 2;
+ hHeaderData->bs_data.limiterBands = 2;
+ hHeaderData->bs_data.limiterGains = 2;
+ hHeaderData->bs_data.interpolFreq = 1;
+ hHeaderData->bs_data.smoothingLength = 1;
+
+ hHeaderData->timeStep = (flags & SBRDEC_ELD_GRID) ? 1 : 2;
+
+ /* Setup pointers to frequency band tables */
+ hFreq->freqBandTable[0] = hFreq->freqBandTableLo;
+ hFreq->freqBandTable[1] = hFreq->freqBandTableHi;
+
+ /* Patch some entries */
+ if (sampleRateOut > 24000) { /* Trigger an error if SBR is going to be processed without */
+ hHeaderData->bs_data.startFreq = 7; /* having read these frequency values from bit stream before. */
+ hHeaderData->bs_data.stopFreq = 3;
+ }
+
+ /* One SBR timeslot corresponds to the amount of samples equal to the amount of analysis bands, divided by the timestep. */
+ hHeaderData->numberTimeSlots = (samplesPerFrame/numAnalysisBands) >> (hHeaderData->timeStep - 1);
+ if (hHeaderData->numberTimeSlots > (16)) {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ hHeaderData->numberOfAnalysisBands = numAnalysisBands;
+
+bail:
+ return sbrError;
+}
+
+
+/*!
+ \brief Initialize the SBR_PREV_FRAME_DATA struct
+*/
+void
+initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data, /*!< handle to struct SBR_PREV_FRAME_DATA */
+ int timeSlots) /*!< Framelength in SBR-timeslots */
+{
+ int i;
+
+ /* Set previous energy and noise levels to 0 for the case
+ that decoding starts in the middle of a bitstream */
+ for (i=0; i < MAX_FREQ_COEFFS; i++)
+ h_prev_data->sfb_nrg_prev[i] = (FIXP_DBL)0;
+ for (i=0; i < MAX_NOISE_COEFFS; i++)
+ h_prev_data->prevNoiseLevel[i] = (FIXP_DBL)0;
+ for (i=0; i < MAX_INVF_BANDS; i++)
+ h_prev_data->sbr_invf_mode[i] = INVF_OFF;
+
+ h_prev_data->stopPos = timeSlots;
+ h_prev_data->coupling = COUPLING_OFF;
+ h_prev_data->ampRes = 0;
+}
+
+
+/*!
+ \brief Read header data from bitstream
+
+ \return error status - 0 if ok
+*/
+SBR_HEADER_STATUS
+sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_FDK_BITSTREAM hBs,
+ const UINT flags,
+ const int fIsSbrData)
+{
+ SBR_HEADER_DATA_BS *pBsData;
+ SBR_HEADER_DATA_BS lastHeader;
+ SBR_HEADER_DATA_BS_INFO lastInfo;
+ int headerExtra1=0, headerExtra2=0;
+
+ /* Copy SBR bit stream header to temporary header */
+ lastHeader = hHeaderData->bs_data;
+ lastInfo = hHeaderData->bs_info;
+
+ /* Read new header from bitstream */
+ {
+ pBsData = &hHeaderData->bs_data;
+ }
+
+ {
+ hHeaderData->bs_info.ampResolution = FDKreadBits (hBs, 1);
+ }
+
+ pBsData->startFreq = FDKreadBits (hBs, 4);
+ pBsData->stopFreq = FDKreadBits (hBs, 4);
+
+ {
+ hHeaderData->bs_info.xover_band = FDKreadBits (hBs, 3);
+ FDKreadBits (hBs, 2);
+ }
+
+ headerExtra1 = FDKreadBits (hBs, 1);
+ headerExtra2 = FDKreadBits (hBs, 1);
+
+ /* Handle extra header information */
+ if( headerExtra1)
+ {
+ pBsData->freqScale = FDKreadBits (hBs, 2);
+ pBsData->alterScale = FDKreadBits (hBs, 1);
+ pBsData->noise_bands = FDKreadBits (hBs, 2);
+ }
+ else {
+ pBsData->freqScale = 2;
+ pBsData->alterScale = 1;
+ pBsData->noise_bands = 2;
+ }
+
+ if (headerExtra2) {
+ pBsData->limiterBands = FDKreadBits (hBs, 2);
+ pBsData->limiterGains = FDKreadBits (hBs, 2);
+ pBsData->interpolFreq = FDKreadBits (hBs, 1);
+ pBsData->smoothingLength = FDKreadBits (hBs, 1);
+ }
+ else {
+ pBsData->limiterBands = 2;
+ pBsData->limiterGains = 2;
+ pBsData->interpolFreq = 1;
+ pBsData->smoothingLength = 1;
+ }
+
+ /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */
+ if(hHeaderData->syncState != SBR_ACTIVE ||
+ lastHeader.startFreq != pBsData->startFreq ||
+ lastHeader.stopFreq != pBsData->stopFreq ||
+ lastHeader.freqScale != pBsData->freqScale ||
+ lastHeader.alterScale != pBsData->alterScale ||
+ lastHeader.noise_bands != pBsData->noise_bands ||
+ lastInfo.xover_band != hHeaderData->bs_info.xover_band) {
+ return HEADER_RESET; /* New settings */
+ }
+
+ return HEADER_OK;
+}
+
+/*!
+ \brief Get missing harmonics parameters (only used for AAC+SBR)
+
+ \return error status - 0 if ok
+*/
+int
+sbrGetSyntheticCodedData(HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameData,
+ HANDLE_FDK_BITSTREAM hBs)
+{
+ int i, bitsRead = 0;
+
+ int flag = FDKreadBits(hBs,1);
+ bitsRead++;
+
+ if(flag){
+ for(i=0;i<hHeaderData->freqBandData.nSfb[1];i++){
+ hFrameData->addHarmonics[i] = FDKreadBits (hBs, 1 );
+ bitsRead++;
+ }
+ }
+ else {
+ for(i=0; i<MAX_FREQ_COEFFS; i++)
+ hFrameData->addHarmonics[i] = 0;
+ }
+ return(bitsRead);
+}
+
+/*!
+ \brief Reads extension data from the bitstream
+
+ The bitstream format allows up to 4 kinds of extended data element.
+ Extended data may contain several elements, each identified by a 2-bit-ID.
+ So far, no extended data elements are defined hence the first 2 parameters
+ are unused. The data should be skipped in order to update the number
+ of read bits for the consistency check in applySBR().
+*/
+static int extractExtendedData(
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< handle to SBR header */
+ HANDLE_FDK_BITSTREAM hBs /*!< Handle to the bit buffer */
+ ,HANDLE_PS_DEC hParametricStereoDec /*!< Parametric Stereo Decoder */
+ ) {
+ INT nBitsLeft;
+ int extended_data;
+ int i, frameOk = 1;
+
+
+ extended_data = FDKreadBits(hBs, 1);
+
+ if (extended_data) {
+ int cnt;
+ int bPsRead = 0;
+
+ cnt = FDKreadBits(hBs, 4);
+ if (cnt == (1<<4)-1)
+ cnt += FDKreadBits(hBs, 8);
+
+
+ nBitsLeft = 8 * cnt;
+
+ /* sanity check for cnt */
+ if (nBitsLeft > (INT)FDKgetValidBits(hBs)) {
+ /* limit nBitsLeft */
+ nBitsLeft = (INT)FDKgetValidBits(hBs);
+ /* set frame error */
+ frameOk = 0;
+ }
+
+ while (nBitsLeft > 7) {
+ int extension_id = FDKreadBits(hBs, 2);
+ nBitsLeft -= 2;
+
+ switch(extension_id) {
+
+
+
+ case EXTENSION_ID_PS_CODING:
+
+ /* Read PS data from bitstream */
+
+ if (hParametricStereoDec != NULL) {
+ if(bPsRead && !hParametricStereoDec->bsData[hParametricStereoDec->bsReadSlot].mpeg.bPsHeaderValid) {
+ cnt = nBitsLeft >> 3; /* number of remaining bytes */
+ for (i=0; i<cnt; i++)
+ FDKreadBits(hBs, 8);
+ nBitsLeft -= cnt * 8;
+ } else {
+ nBitsLeft -= ReadPsData(hParametricStereoDec, hBs, nBitsLeft);
+ bPsRead = 1;
+ }
+ }
+
+ /* parametric stereo detected, could set channelMode accordingly here */
+ /* */
+ /* "The usage of this parametric stereo extension to HE-AAC is */
+ /* signalled implicitly in the bitstream. Hence, if an sbr_extension() */
+ /* with bs_extension_id==EXTENSION_ID_PS is found in the SBR part of */
+ /* the bitstream, a decoder supporting the combination of SBR and PS */
+ /* shall operate the PS tool to generate a stereo output signal." */
+ /* source: ISO/IEC 14496-3:2001/FDAM 2:2004(E) */
+
+ break;
+
+
+ default:
+ cnt = nBitsLeft >> 3; /* number of remaining bytes */
+ for (i=0; i<cnt; i++)
+ FDKreadBits(hBs, 8);
+ nBitsLeft -= cnt * 8;
+ break;
+ }
+ }
+
+ if (nBitsLeft < 0) {
+ frameOk = 0;
+ goto bail;
+ }
+ else {
+ /* Read fill bits for byte alignment */
+ FDKreadBits(hBs, nBitsLeft);
+ }
+ }
+
+bail:
+ return (frameOk);
+}
+
+
+/*!
+ \brief Read bitstream elements of one channel
+
+ \return SbrFrameOK: 1=ok, 0=error
+*/
+int
+sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */
+ HANDLE_FDK_BITSTREAM hBs, /*!< Handle to struct BIT_BUF */
+ HANDLE_PS_DEC hParametricStereoDec, /*!< Handle to PS decoder */
+ const UINT flags,
+ const int overlap
+ )
+{
+ int i;
+
+
+ hFrameData->coupling = COUPLING_OFF;
+
+ {
+ /* Reserved bits */
+ if (FDKreadBits(hBs, 1)) { /* bs_data_extra */
+ FDKreadBits(hBs, 4);
+ if (flags & SBRDEC_SYNTAX_SCAL) {
+ FDKreadBits(hBs, 4);
+ }
+ }
+ }
+
+ if (flags & SBRDEC_SYNTAX_SCAL) {
+ FDKreadBits (hBs, 1); /* bs_coupling */
+ }
+
+ /*
+ Grid control
+ */
+ if ( !extractFrameInfo ( hBs, hHeaderData, hFrameData, 1, flags) )
+ return 0;
+
+ if ( !checkFrameInfo (&hFrameData->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+ return 0;
+
+
+ /*
+ Fetch domain vectors (time or frequency direction for delta-coding)
+ */
+ sbrGetDirectionControlData (hFrameData, hBs);
+
+ for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+ hFrameData->sbr_invf_mode[i] =
+ (INVF_MODE) FDKreadBits (hBs, 2);
+ }
+
+
+
+ /* raw data */
+ if ( !sbrGetEnvelope (hHeaderData, hFrameData, hBs, flags) )
+ return 0;
+
+
+ sbrGetNoiseFloorData (hHeaderData, hFrameData, hBs);
+
+ sbrGetSyntheticCodedData(hHeaderData, hFrameData, hBs);
+
+ {
+ /* sbr extended data */
+ if (! extractExtendedData(
+ hHeaderData,
+ hBs
+ ,hParametricStereoDec
+ )) {
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+
+
+/*!
+ \brief Read bitstream elements of a channel pair
+ \return SbrFrameOK
+*/
+int
+sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA hFrameDataLeft, /*!< Dynamic control data for first channel */
+ HANDLE_SBR_FRAME_DATA hFrameDataRight,/*!< Dynamic control data for second channel */
+ HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */
+ const UINT flags,
+ const int overlap )
+{
+ int i, bit;
+
+
+ /* Reserved bits */
+ if (FDKreadBits(hBs, 1)) { /* bs_data_extra */
+ FDKreadBits(hBs, 4);
+ FDKreadBits(hBs, 4);
+ }
+
+ /* Read coupling flag */
+ bit = FDKreadBits (hBs, 1);
+
+ if (bit) {
+ hFrameDataLeft->coupling = COUPLING_LEVEL;
+ hFrameDataRight->coupling = COUPLING_BAL;
+ }
+ else {
+ hFrameDataLeft->coupling = COUPLING_OFF;
+ hFrameDataRight->coupling = COUPLING_OFF;
+ }
+
+
+ /*
+ Grid control
+ */
+ if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataLeft, 2, flags) )
+ return 0;
+
+ if ( !checkFrameInfo (&hFrameDataLeft->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+ return 0;
+
+ if (hFrameDataLeft->coupling) {
+ FDKmemcpy (&hFrameDataRight->frameInfo, &hFrameDataLeft->frameInfo, sizeof(FRAME_INFO));
+ hFrameDataRight->ampResolutionCurrentFrame = hFrameDataLeft->ampResolutionCurrentFrame;
+ }
+ else {
+ if ( !extractFrameInfo (hBs, hHeaderData, hFrameDataRight, 2, flags) )
+ return 0;
+
+ if ( !checkFrameInfo (&hFrameDataRight->frameInfo, hHeaderData->numberTimeSlots, overlap, hHeaderData->timeStep) )
+ return 0;
+ }
+
+ /*
+ Fetch domain vectors (time or frequency direction for delta-coding)
+ */
+ sbrGetDirectionControlData (hFrameDataLeft, hBs);
+ sbrGetDirectionControlData (hFrameDataRight, hBs);
+
+ for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+ hFrameDataLeft->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2);
+ }
+
+ if (hFrameDataLeft->coupling) {
+ for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+ hFrameDataRight->sbr_invf_mode[i] = hFrameDataLeft->sbr_invf_mode[i];
+ }
+
+
+ if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) ) {
+ return 0;
+ }
+
+ sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs);
+
+ if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) ) {
+ return 0;
+ }
+ }
+ else {
+
+ for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+ hFrameDataRight->sbr_invf_mode[i] = (INVF_MODE) FDKreadBits (hBs, 2);
+ }
+
+
+
+ if ( !sbrGetEnvelope (hHeaderData, hFrameDataLeft, hBs, flags) )
+ return 0;
+
+ if ( !sbrGetEnvelope (hHeaderData, hFrameDataRight, hBs, flags) )
+ return 0;
+
+ sbrGetNoiseFloorData (hHeaderData, hFrameDataLeft, hBs);
+
+ }
+ sbrGetNoiseFloorData (hHeaderData, hFrameDataRight, hBs);
+
+ sbrGetSyntheticCodedData(hHeaderData, hFrameDataLeft, hBs);
+ sbrGetSyntheticCodedData(hHeaderData, hFrameDataRight, hBs);
+
+ {
+ if (! extractExtendedData(
+ hHeaderData,
+ hBs
+ ,NULL
+ ) ) {
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+
+
+
+/*!
+ \brief Read direction control data from bitstream
+*/
+void
+sbrGetDirectionControlData (HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+ HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */
+{
+ int i;
+
+ for (i = 0; i < h_frame_data->frameInfo.nEnvelopes; i++) {
+ h_frame_data->domain_vec[i] = FDKreadBits (hBs, 1);
+ }
+
+ for (i = 0; i < h_frame_data->frameInfo.nNoiseEnvelopes; i++) {
+ h_frame_data->domain_vec_noise[i] = FDKreadBits (hBs, 1);
+ }
+}
+
+
+
+/*!
+ \brief Read noise-floor-level data from bitstream
+*/
+void
+sbrGetNoiseFloorData (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+ HANDLE_FDK_BITSTREAM hBs) /*!< handle to struct BIT_BUF */
+{
+ int i,j;
+ int delta;
+ COUPLING_MODE coupling;
+ int noNoiseBands = hHeaderData->freqBandData.nNfb;
+
+ Huffman hcb_noiseF;
+ Huffman hcb_noise;
+ int envDataTableCompFactor;
+
+ coupling = h_frame_data->coupling;
+
+
+ /*
+ Select huffman codebook depending on coupling mode
+ */
+ if (coupling == COUPLING_BAL) {
+ hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T;
+ hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F; /* "sbr_huffBook_NoiseBalance11F" */
+ envDataTableCompFactor = 1;
+ }
+ else {
+ hcb_noise = (Huffman)&FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T;
+ hcb_noiseF = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F; /* "sbr_huffBook_NoiseLevel11F" */
+ envDataTableCompFactor = 0;
+ }
+
+ /*
+ Read raw noise-envelope data
+ */
+ for (i=0; i<h_frame_data->frameInfo.nNoiseEnvelopes; i++) {
+
+
+ if (h_frame_data->domain_vec_noise[i] == 0) {
+ if (coupling == COUPLING_BAL) {
+ h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] =
+ (FIXP_SGL) (((int)FDKreadBits (hBs, 5)) << envDataTableCompFactor);
+ }
+ else {
+ h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands] =
+ (FIXP_SGL) (int)FDKreadBits (hBs, 5);
+ }
+
+ for (j = 1; j < noNoiseBands; j++) {
+ delta = DecodeHuffmanCW(hcb_noiseF, hBs);
+ h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor);
+ }
+ }
+ else {
+ for (j = 0; j < noNoiseBands; j++) {
+ delta = DecodeHuffmanCW(hcb_noise, hBs);
+ h_frame_data->sbrNoiseFloorLevel[i*noNoiseBands+j] = (FIXP_SGL) (delta << envDataTableCompFactor);
+ }
+ }
+ }
+}
+
+
+/*!
+ \brief Read envelope data from bitstream
+*/
+static int
+sbrGetEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< handle to struct SBR_FRAME_DATA */
+ HANDLE_FDK_BITSTREAM hBs, /*!< handle to struct BIT_BUF */
+ const UINT flags)
+{
+ int i, j;
+ UCHAR no_band[MAX_ENVELOPES];
+ int delta = 0;
+ int offset = 0;
+ COUPLING_MODE coupling = h_frame_data->coupling;
+ int ampRes = hHeaderData->bs_info.ampResolution;
+ int nEnvelopes = h_frame_data->frameInfo.nEnvelopes;
+ int envDataTableCompFactor;
+ int start_bits, start_bits_balance;
+ Huffman hcb_t, hcb_f;
+
+ h_frame_data->nScaleFactors = 0;
+
+ if ( (h_frame_data->frameInfo.frameClass == 0) && (nEnvelopes == 1) ) {
+ if (flags & SBRDEC_ELD_GRID)
+ ampRes = h_frame_data->ampResolutionCurrentFrame;
+ else
+ ampRes = 0;
+ }
+ h_frame_data->ampResolutionCurrentFrame = ampRes;
+
+ /*
+ Set number of bits for first value depending on amplitude resolution
+ */
+ if(ampRes == 1)
+ {
+ start_bits = 6;
+ start_bits_balance = 5;
+ }
+ else
+ {
+ start_bits = 7;
+ start_bits_balance = 6;
+ }
+
+ /*
+ Calculate number of values for each envelope and alltogether
+ */
+ for (i = 0; i < nEnvelopes; i++) {
+ no_band[i] = hHeaderData->freqBandData.nSfb[h_frame_data->frameInfo.freqRes[i]];
+ h_frame_data->nScaleFactors += no_band[i];
+ }
+ if (h_frame_data->nScaleFactors > MAX_NUM_ENVELOPE_VALUES)
+ return 0;
+
+ /*
+ Select Huffman codebook depending on coupling mode and amplitude resolution
+ */
+ if (coupling == COUPLING_BAL) {
+ envDataTableCompFactor = 1;
+ if (ampRes == 0) {
+ hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10T;
+ hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance10F;
+ }
+ else {
+ hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11T;
+ hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvBalance11F;
+ }
+ }
+ else {
+ envDataTableCompFactor = 0;
+ if (ampRes == 0) {
+ hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10T;
+ hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel10F;
+ }
+ else {
+ hcb_t = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11T;
+ hcb_f = (Huffman)&FDK_sbrDecoder_sbr_huffBook_EnvLevel11F;
+ }
+ }
+
+ /*
+ Now read raw envelope data
+ */
+ for (j = 0, offset = 0; j < nEnvelopes; j++) {
+
+
+ if (h_frame_data->domain_vec[j] == 0) {
+ if (coupling == COUPLING_BAL) {
+ h_frame_data->iEnvelope[offset] =
+ (FIXP_SGL) (( (int)FDKreadBits(hBs, start_bits_balance)) << envDataTableCompFactor);
+ }
+ else {
+ h_frame_data->iEnvelope[offset] =
+ (FIXP_SGL) (int)FDKreadBits (hBs, start_bits);
+ }
+ }
+
+ for (i = (1 - h_frame_data->domain_vec[j]); i < no_band[j]; i++) {
+
+ if (h_frame_data->domain_vec[j] == 0) {
+ delta = DecodeHuffmanCW(hcb_f, hBs);
+ }
+ else {
+ delta = DecodeHuffmanCW(hcb_t, hBs);
+ }
+
+ h_frame_data->iEnvelope[offset + i] = (FIXP_SGL) (delta << envDataTableCompFactor);
+ }
+ offset += no_band[j];
+ }
+
+#if ENV_EXP_FRACT
+ /* Convert from int to scaled fract (ENV_EXP_FRACT bits for the fractional part) */
+ for (i = 0; i < h_frame_data->nScaleFactors; i++) {
+ h_frame_data->iEnvelope[i] <<= ENV_EXP_FRACT;
+ }
+#endif
+
+ return 1;
+}
+
+
+//static const FRAME_INFO v_frame_info1_8 = { 0, 1, {0, 8}, {1}, -1, 1, {0, 8} };
+static const FRAME_INFO v_frame_info2_8 = { 0, 2, {0, 4, 8}, {1, 1}, -1, 2, {0, 4, 8} };
+static const FRAME_INFO v_frame_info4_8 = { 0, 4, {0, 2, 4, 6, 8}, {1, 1, 1, 1}, -1, 2, {0, 4, 8} };
+
+/***************************************************************************/
+/*!
+ \brief Generates frame info for FIXFIXonly frame class used for low delay version
+
+ \return nothing
+ ****************************************************************************/
+ static void generateFixFixOnly ( FRAME_INFO *hSbrFrameInfo,
+ int tranPosInternal,
+ int numberTimeSlots
+ )
+{
+ int nEnv, i, tranIdx;
+ const int *pTable;
+
+ switch (numberTimeSlots) {
+ case 8:
+ pTable = FDK_sbrDecoder_envelopeTable_8[tranPosInternal];
+ break;
+ case 15:
+ pTable = FDK_sbrDecoder_envelopeTable_15[tranPosInternal];
+ break;
+ case 16:
+ pTable = FDK_sbrDecoder_envelopeTable_16[tranPosInternal];
+ break;
+ default:
+ FDK_ASSERT(0);
+ }
+
+ /* look number of envelopes in table */
+ nEnv = pTable[0];
+ /* look up envelope distribution in table */
+ for (i=1; i<nEnv; i++)
+ hSbrFrameInfo->borders[i] = pTable[i+2];
+ /* open and close frame border */
+ hSbrFrameInfo->borders[0] = 0;
+ hSbrFrameInfo->borders[nEnv] = numberTimeSlots;
+ hSbrFrameInfo->nEnvelopes = nEnv;
+
+ /* transient idx */
+ tranIdx = hSbrFrameInfo->tranEnv = pTable[1];
+
+ /* add noise floors */
+ hSbrFrameInfo->bordersNoise[0] = 0;
+ hSbrFrameInfo->bordersNoise[1] = hSbrFrameInfo->borders[tranIdx?tranIdx:1];
+ hSbrFrameInfo->bordersNoise[2] = numberTimeSlots;
+ /* nEnv is always > 1, so nNoiseEnvelopes is always 2 (IEC 14496-3 4.6.19.3.2) */
+ hSbrFrameInfo->nNoiseEnvelopes = 2;
+}
+
+/*!
+ \brief Extracts LowDelaySBR control data from the bitstream.
+
+ \return zero for bitstream error, one for correct.
+*/
+static int
+extractLowDelayGrid (HANDLE_FDK_BITSTREAM hBitBuf, /*!< bitbuffer handle */
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< contains the FRAME_INFO struct to be filled */
+ int timeSlots
+ )
+{
+ FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo;
+ INT numberTimeSlots = hHeaderData->numberTimeSlots;
+ INT temp = 0, k;
+
+ /* FIXFIXonly framing case */
+ h_frame_data->frameInfo.frameClass = 0;
+
+ /* get the transient position from the bitstream */
+ switch (timeSlots){
+ case 8:
+ /* 3bit transient position (temp={0;..;7}) */
+ temp = FDKreadBits( hBitBuf, 3);
+ break;
+
+ case 16:
+ case 15:
+ /* 4bit transient position (temp={0;..;15}) */
+ temp = FDKreadBits( hBitBuf, 4);
+ break;
+
+ default:
+ return 0;
+ }
+
+ /* calculate borders according to the transient position */
+ generateFixFixOnly ( pFrameInfo,
+ temp,
+ numberTimeSlots
+ );
+
+ /* decode freq res: */
+ for (k = 0; k < pFrameInfo->nEnvelopes; k++) {
+ pFrameInfo->freqRes[k] = (UCHAR) FDKreadBits (hBitBuf, 1); /* f = F [1 bits] */
+ }
+
+
+ return 1;
+}
+
+/*!
+ \brief Extract the frame information (structure FRAME_INFO) from the bitstream
+ \return Zero for bitstream error, one for correct.
+*/
+int
+extractFrameInfo ( HANDLE_FDK_BITSTREAM hBs, /*!< bitbuffer handle */
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ HANDLE_SBR_FRAME_DATA h_frame_data, /*!< pointer to memory where the frame-info will be stored */
+ const UINT nrOfChannels,
+ const UINT flags
+ )
+{
+ FRAME_INFO * pFrameInfo = &h_frame_data->frameInfo;
+ int numberTimeSlots = hHeaderData->numberTimeSlots;
+ int pointer_bits = 0, nEnv = 0, b = 0, border, i, n = 0,
+ k, p, aL, aR, nL, nR,
+ temp = 0, staticFreqRes;
+ UCHAR frameClass;
+
+ if (flags & SBRDEC_ELD_GRID) {
+ /* CODEC_AACLD (LD+SBR) only uses the normal 0 Grid for non-transient Frames and the LowDelayGrid for transient Frames */
+ frameClass = FDKreadBits (hBs, 1); /* frameClass = [1 bit] */
+ if ( frameClass == 1 ) {
+ /* if frameClass == 1, extract LowDelaySbrGrid, otherwise extract normal SBR-Grid for FIXIFX */
+ /* extract the AACLD-Sbr-Grid */
+ pFrameInfo->frameClass = frameClass;
+ extractLowDelayGrid (hBs, hHeaderData, h_frame_data, numberTimeSlots);
+ return 1;
+ }
+ } else
+ {
+ frameClass = FDKreadBits (hBs, 2); /* frameClass = C [2 bits] */
+ }
+
+
+ switch (frameClass) {
+ case 0:
+ temp = FDKreadBits (hBs, 2); /* E [2 bits ] */
+ nEnv = (int) (1 << temp); /* E -> e */
+
+ if ((flags & SBRDEC_ELD_GRID) && (nEnv == 1))
+ h_frame_data->ampResolutionCurrentFrame = FDKreadBits( hBs, 1); /* new ELD Syntax 07-11-09 */
+
+ staticFreqRes = FDKreadBits (hBs, 1);
+
+ {
+ if (nEnv > MAX_ENVELOPES_HEAAC)
+ return 0;
+ }
+
+ b = nEnv + 1;
+ switch (nEnv) {
+ case 1:
+ switch (numberTimeSlots) {
+ case 15:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_15, sizeof(FRAME_INFO));
+ break;
+ case 16:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info1_16, sizeof(FRAME_INFO));
+ break;
+ default:
+ FDK_ASSERT(0);
+ }
+ break;
+ case 2:
+ switch (numberTimeSlots) {
+ case 15:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_15, sizeof(FRAME_INFO));
+ break;
+ case 16:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info2_16, sizeof(FRAME_INFO));
+ break;
+ default:
+ FDK_ASSERT(0);
+ }
+ break;
+ case 4:
+ switch (numberTimeSlots) {
+ case 15:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_15, sizeof(FRAME_INFO));
+ break;
+ case 16:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info4_16, sizeof(FRAME_INFO));
+ break;
+ default:
+ FDK_ASSERT(0);
+ }
+ break;
+ case 8:
+#if (MAX_ENVELOPES >= 8)
+ switch (numberTimeSlots) {
+ case 15:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_15, sizeof(FRAME_INFO));
+ break;
+ case 16:
+ FDKmemcpy (pFrameInfo, &FDK_sbrDecoder_sbr_frame_info8_16, sizeof(FRAME_INFO));
+ break;
+ default:
+ FDK_ASSERT(0);
+ }
+ break;
+#else
+ return 0;
+#endif
+ }
+ /* Apply correct freqRes (High is default) */
+ if (!staticFreqRes) {
+ for (i = 0; i < nEnv ; i++)
+ pFrameInfo->freqRes[i] = 0;
+ }
+
+ break;
+ case 1:
+ case 2:
+ temp = FDKreadBits (hBs, 2); /* A [2 bits] */
+
+ n = FDKreadBits (hBs, 2); /* n = N [2 bits] */
+
+ nEnv = n + 1; /* # envelopes */
+ b = nEnv + 1; /* # borders */
+
+ break;
+ }
+
+ switch (frameClass) {
+ case 1:
+ /* Decode borders: */
+ pFrameInfo->borders[0] = 0; /* first border */
+ border = temp + numberTimeSlots; /* A -> aR */
+ i = b-1; /* frame info index for last border */
+ pFrameInfo->borders[i] = border; /* last border */
+
+ for (k = 0; k < n; k++) {
+ temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+ border -= (2 * temp + 2); /* R -> r */
+ pFrameInfo->borders[--i] = border;
+ }
+
+
+ /* Decode pointer: */
+ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1));
+ p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */
+
+ if (p > n+1)
+ return 0;
+
+ pFrameInfo->tranEnv = p ? n + 2 - p : -1;
+
+
+ /* Decode freq res: */
+ for (k = n; k >= 0; k--) {
+ pFrameInfo->freqRes[k] = FDKreadBits (hBs, 1); /* f = F [1 bits] */
+ }
+
+
+ /* Calculate noise floor middle border: */
+ if (p == 0 || p == 1)
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n];
+ else
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+
+ break;
+
+ case 2:
+ /* Decode borders: */
+ border = temp; /* A -> aL */
+ pFrameInfo->borders[0] = border; /* first border */
+
+ for (k = 1; k <= n; k++) {
+ temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+ border += (2 * temp + 2); /* R -> r */
+ pFrameInfo->borders[k] = border;
+ }
+ pFrameInfo->borders[k] = numberTimeSlots; /* last border */
+
+
+ /* Decode pointer: */
+ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(n+1));
+ p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */
+ if (p > n+1)
+ return 0;
+
+ if (p == 0 || p == 1)
+ pFrameInfo->tranEnv = -1;
+ else
+ pFrameInfo->tranEnv = p - 1;
+
+
+
+ /* Decode freq res: */
+ for (k = 0; k <= n; k++) {
+ pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */
+ }
+
+
+
+ /* Calculate noise floor middle border: */
+ switch (p) {
+ case 0:
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[1];
+ break;
+ case 1:
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[n];
+ break;
+ default:
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+ break;
+ }
+
+ break;
+
+ case 3:
+ /* v_ctrlSignal = [frameClass,aL,aR,nL,nR,v_rL,v_rR,p,v_fLR]; */
+
+ aL = FDKreadBits (hBs, 2); /* AL [2 bits], AL -> aL */
+
+ aR = FDKreadBits (hBs, 2) + numberTimeSlots; /* AR [2 bits], AR -> aR */
+
+ nL = FDKreadBits (hBs, 2); /* nL = NL [2 bits] */
+
+ nR = FDKreadBits (hBs, 2); /* nR = NR [2 bits] */
+
+
+
+ /*-------------------------------------------------------------------------
+ Calculate help variables
+ --------------------------------------------------------------------------*/
+
+ /* general: */
+ nEnv = nL + nR + 1; /* # envelopes */
+ if (nEnv > MAX_ENVELOPES)
+ return 0;
+ b = nEnv + 1; /* # borders */
+
+
+
+ /*-------------------------------------------------------------------------
+ Decode envelopes
+ --------------------------------------------------------------------------*/
+
+
+ /* L-borders: */
+ border = aL; /* first border */
+ pFrameInfo->borders[0] = border;
+
+ for (k = 1; k <= nL; k++) {
+ temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+ border += (2 * temp + 2); /* R -> r */
+ pFrameInfo->borders[k] = border;
+ }
+
+
+ /* R-borders: */
+ border = aR; /* last border */
+ i = nEnv;
+
+ pFrameInfo->borders[i] = border;
+
+ for (k = 0; k < nR; k++) {
+ temp = FDKreadBits (hBs, 2);/* R [2 bits] */
+ border -= (2 * temp + 2); /* R -> r */
+ pFrameInfo->borders[--i] = border;
+ }
+
+
+ /* decode pointer: */
+ pointer_bits = DFRACT_BITS - 1 - CountLeadingBits((FIXP_DBL)(nL+nR+1));
+ p = FDKreadBits (hBs, pointer_bits); /* p = P [pointer_bits bits] */
+
+ if (p > nL+nR+1)
+ return 0;
+
+ pFrameInfo->tranEnv = p ? b - p : -1;
+
+
+
+ /* decode freq res: */
+ for (k = 0; k < nEnv; k++) {
+ pFrameInfo->freqRes[k] = FDKreadBits(hBs, 1); /* f = F [1 bits] */
+ }
+
+
+
+ /*-------------------------------------------------------------------------
+ Decode noise floors
+ --------------------------------------------------------------------------*/
+ pFrameInfo->bordersNoise[0] = aL;
+
+ if (nEnv == 1) {
+ /* 1 noise floor envelope: */
+ pFrameInfo->bordersNoise[1] = aR;
+ }
+ else {
+ /* 2 noise floor envelopes */
+ if (p == 0 || p == 1)
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[nEnv - 1];
+ else
+ pFrameInfo->bordersNoise[1] = pFrameInfo->borders[pFrameInfo->tranEnv];
+ pFrameInfo->bordersNoise[2] = aR;
+ }
+ break;
+ }
+
+
+ /*
+ Store number of envelopes, noise floor envelopes and frame class
+ */
+ pFrameInfo->nEnvelopes = nEnv;
+
+ if (nEnv == 1)
+ pFrameInfo->nNoiseEnvelopes = 1;
+ else
+ pFrameInfo->nNoiseEnvelopes = 2;
+
+ pFrameInfo->frameClass = frameClass;
+
+ if (pFrameInfo->frameClass == 2 || pFrameInfo->frameClass == 1) {
+ /* calculate noise floor first and last borders: */
+ pFrameInfo->bordersNoise[0] = pFrameInfo->borders[0];
+ pFrameInfo->bordersNoise[pFrameInfo->nNoiseEnvelopes] = pFrameInfo->borders[nEnv];
+ }
+
+
+ return 1;
+}
+
+
+/*!
+ \brief Check if the frameInfo vector has reasonable values.
+ \return Zero for error, one for correct
+*/
+static int
+checkFrameInfo (FRAME_INFO * pFrameInfo, /*!< pointer to frameInfo */
+ int numberOfTimeSlots, /*!< QMF time slots per frame */
+ int overlap, /*!< Amount of overlap QMF time slots */
+ int timeStep) /*!< QMF slots to SBR slots step factor */
+{
+ int maxPos,i,j;
+ int startPos;
+ int stopPos;
+ int tranEnv;
+ int startPosNoise;
+ int stopPosNoise;
+ int nEnvelopes = pFrameInfo->nEnvelopes;
+ int nNoiseEnvelopes = pFrameInfo->nNoiseEnvelopes;
+
+ if(nEnvelopes < 1 || nEnvelopes > MAX_ENVELOPES)
+ return 0;
+
+ if(nNoiseEnvelopes > MAX_NOISE_ENVELOPES)
+ return 0;
+
+ startPos = pFrameInfo->borders[0];
+ stopPos = pFrameInfo->borders[nEnvelopes];
+ tranEnv = pFrameInfo->tranEnv;
+ startPosNoise = pFrameInfo->bordersNoise[0];
+ stopPosNoise = pFrameInfo->bordersNoise[nNoiseEnvelopes];
+
+ if (overlap < 0 || overlap > (6)) {
+ return 0;
+ }
+ if (timeStep < 1 || timeStep > 2) {
+ return 0;
+ }
+ maxPos = numberOfTimeSlots + (overlap/timeStep);
+
+ /* Check that the start and stop positions of the frame are reasonable values. */
+ if( (startPos < 0) || (startPos >= stopPos) )
+ return 0;
+ if( startPos > maxPos-numberOfTimeSlots ) /* First env. must start in or directly after the overlap buffer */
+ return 0;
+ if( stopPos < numberOfTimeSlots ) /* One complete frame must be ready for output after processing */
+ return 0;
+ if(stopPos > maxPos)
+ return 0;
+
+ /* Check that the start border for every envelope is strictly later in time */
+ for(i=0;i<nEnvelopes;i++) {
+ if(pFrameInfo->borders[i] >= pFrameInfo->borders[i+1])
+ return 0;
+ }
+
+ /* Check that the envelope to be shortened is actually among the envelopes */
+ if(tranEnv>nEnvelopes)
+ return 0;
+
+
+ /* Check the noise borders */
+ if(nEnvelopes==1 && nNoiseEnvelopes>1)
+ return 0;
+
+ if(startPos != startPosNoise || stopPos != stopPosNoise)
+ return 0;
+
+
+ /* Check that the start border for every noise-envelope is strictly later in time*/
+ for(i=0; i<nNoiseEnvelopes; i++) {
+ if(pFrameInfo->bordersNoise[i] >= pFrameInfo->bordersNoise[i+1])
+ return 0;
+ }
+
+ /* Check that every noise border is the same as an envelope border*/
+ for(i=0; i<nNoiseEnvelopes; i++) {
+ startPosNoise = pFrameInfo->bordersNoise[i];
+
+ for(j=0; j<nEnvelopes; j++) {
+ if(pFrameInfo->borders[j] == startPosNoise)
+ break;
+ }
+ if(j==nEnvelopes)
+ return 0;
+ }
+
+ return 1;
+}
diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h
new file mode 100644
index 0000000..be46246
--- /dev/null
+++ b/libSBRdec/src/env_extr.h
@@ -0,0 +1,319 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Envelope extraction prototypes
+*/
+
+#ifndef __ENVELOPE_EXTRACTION_H
+#define __ENVELOPE_EXTRACTION_H
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+#include "lpp_tran.h"
+
+#include "psdec.h"
+
+#define ENV_EXP_FRACT 0
+/*!< Shift raw envelope data to support fractional numbers.
+ Can be set to 8 instead of 0 to enhance accuracy during concealment.
+ This is not required for conformance and #requantizeEnvelopeData() will
+ become more expensive.
+*/
+
+#define EXP_BITS 6
+/*!< Size of exponent-part of a pseudo float envelope value (should be at least 6).
+ The remaining bits in each word are used for the mantissa (should be at least 10).
+ This format is used in the arrays iEnvelope[] and sbrNoiseFloorLevel[]
+ in the FRAME_DATA struct which must fit in a certain part of the output buffer
+ (See buffer management in sbr_dec.cpp).
+ Exponents and mantissas could also be stored in separate arrays.
+ Accessing the exponent or the mantissa would be simplified and the masks #MASK_E
+ resp. #MASK_M would no longer be required.
+*/
+
+#define MASK_M (((1 << (FRACT_BITS - EXP_BITS)) - 1) << EXP_BITS) /*!< Mask for extracting the mantissa of a pseudo float envelope value */
+#define MASK_E ((1 << EXP_BITS) - 1) /*!< Mask for extracting the exponent of a pseudo float envelope value */
+
+#define SIGN_EXT ( ((SCHAR)-1) ^ MASK_E) /*!< a CHAR-constant with all bits above our sign-bit set */
+#define ROUNDING ( (FIXP_SGL)(1<<(EXP_BITS-1)) ) /*!< 0.5-offset for rounding the mantissa of a pseudo-float envelope value */
+#define NRG_EXP_OFFSET 16 /*!< Will be added to the reference energy's exponent to prevent negative numbers */
+#define NOISE_EXP_OFFSET 38 /*!< Will be added to the noise level exponent to prevent negative numbers */
+
+typedef enum
+{
+ HEADER_NOT_PRESENT,
+ HEADER_OK,
+ HEADER_RESET
+}
+SBR_HEADER_STATUS;
+
+typedef enum
+{
+ SBR_NOT_INITIALIZED,
+ UPSAMPLING,
+ SBR_HEADER,
+ SBR_ACTIVE
+}
+SBR_SYNC_STATE;
+
+
+typedef enum
+{
+ COUPLING_OFF = 0,
+ COUPLING_LEVEL,
+ COUPLING_BAL
+}
+COUPLING_MODE;
+
+typedef struct
+{
+ UCHAR nSfb[2]; /*!< Number of SBR-bands for low and high freq-resolution */
+ UCHAR nNfb; /*!< Actual number of noise bands to read from the bitstream*/
+ UCHAR numMaster; /*!< Number of SBR-bands in v_k_master */
+ UCHAR lowSubband; /*!< QMF-band where SBR frequency range starts */
+ UCHAR highSubband; /*!< QMF-band where SBR frequency range ends */
+ UCHAR limiterBandTable[MAX_NUM_LIMITERS+1]; /*!< Limiter band table. */
+ UCHAR noLimiterBands; /*!< Number of limiter bands. */
+ UCHAR nInvfBands; /*!< Number of bands for inverse filtering */
+ UCHAR *freqBandTable[2]; /*!< Pointers to freqBandTableLo and freqBandTableHi */
+ UCHAR freqBandTableLo[MAX_FREQ_COEFFS/2+1];
+ /*!< Mapping of SBR bands to QMF bands for low frequency resolution */
+ UCHAR freqBandTableHi[MAX_FREQ_COEFFS+1];
+ /*!< Mapping of SBR bands to QMF bands for high frequency resolution */
+ UCHAR freqBandTableNoise[MAX_NOISE_COEFFS+1];
+ /*!< Mapping of SBR noise bands to QMF bands */
+ UCHAR v_k_master[MAX_FREQ_COEFFS+1];
+ /*!< Master BandTable which freqBandTable is derived from */
+}
+FREQ_BAND_DATA;
+
+typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
+
+#define SBRDEC_ELD_GRID 1
+#define SBRDEC_SYNTAX_SCAL 2
+#define SBRDEC_SYNTAX_USAC 4
+#define SBRDEC_SYNTAX_RSVD50 8
+#define SBRDEC_LOW_POWER 16 /* Flag indicating that Low Power QMF mode shall be used. */
+#define SBRDEC_PS_DECODED 32 /* Flag indicating that PS was decoded and rendered. */
+#define SBRDEC_LD_MPS_QMF 512 /* Flag indicating that the LD-MPS QMF shall be used. */
+
+#define SBRDEC_HDR_STAT_RESET 1
+#define SBRDEC_HDR_STAT_UPDATE 2
+
+typedef struct {
+ UCHAR ampResolution; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */
+ UCHAR xover_band; /*!< Start index in #v_k_master[] used for dynamic crossover frequency */
+ UCHAR sbr_preprocessing; /*!< SBR prewhitening flag. */
+} SBR_HEADER_DATA_BS_INFO;
+
+typedef struct {
+ /* Changes in these variables causes a reset of the decoder */
+ UCHAR startFreq; /*!< Index for SBR start frequency */
+ UCHAR stopFreq; /*!< Index for SBR highest frequency */
+ UCHAR freqScale; /*!< 0: linear scale, 1-3 logarithmic scales */
+ UCHAR alterScale; /*!< Flag for coarser frequency resolution */
+ UCHAR noise_bands; /*!< Noise bands per octave, read from bitstream*/
+
+ /* don't require reset */
+ UCHAR limiterBands; /*!< Index for number of limiter bands per octave */
+ UCHAR limiterGains; /*!< Index to select gain limit */
+ UCHAR interpolFreq; /*!< Select gain calculation method (1: per QMF channel, 0: per SBR band) */
+ UCHAR smoothingLength; /*!< Smoothing of gains over time (0: on 1: off) */
+
+} SBR_HEADER_DATA_BS;
+
+typedef struct
+{
+ SBR_SYNC_STATE syncState; /*!< The current initialization status of the header */
+
+ UCHAR status; /*!< Flags field used for signaling a reset right before the processing starts and an update from config (e.g. ASC). */
+ UCHAR frameErrorFlag; /*!< Frame data valid flag. CAUTION: This variable will be overwritten by the flag stored in the element structure.
+ This is necessary because of the frame delay. There it might happen that different slots use the same header. */
+ UCHAR numberTimeSlots; /*!< AAC: 16,15 */
+ UCHAR numberOfAnalysisBands; /*!< Number of QMF analysis bands */
+ UCHAR timeStep; /*!< Time resolution of SBR in QMF-slots */
+ UINT sbrProcSmplRate; /*!< SBR processing sampling frequency (!= OutputSamplingRate)
+ (always: CoreSamplingRate * UpSamplingFactor; even in single rate mode) */
+
+ SBR_HEADER_DATA_BS bs_data; /*!< current SBR header. */
+ SBR_HEADER_DATA_BS_INFO bs_info; /*!< SBR info. */
+
+ FREQ_BAND_DATA freqBandData; /*!< Pointer to struct #FREQ_BAND_DATA */
+}
+SBR_HEADER_DATA;
+
+typedef SBR_HEADER_DATA *HANDLE_SBR_HEADER_DATA;
+
+
+typedef struct
+{
+ UCHAR frameClass; /*!< Select grid type */
+ UCHAR nEnvelopes; /*!< Number of envelopes */
+ UCHAR borders[MAX_ENVELOPES+1]; /*!< Envelope borders (in SBR-timeslots, e.g. mp3PRO: 0..11) */
+ UCHAR freqRes[MAX_ENVELOPES]; /*!< Frequency resolution for each envelope (0=low, 1=high) */
+ SCHAR tranEnv; /*!< Transient envelope, -1 if none */
+ UCHAR nNoiseEnvelopes; /*!< Number of noise envelopes */
+ UCHAR bordersNoise[MAX_NOISE_ENVELOPES+1];/*!< borders of noise envelopes */
+}
+FRAME_INFO;
+
+
+typedef struct
+{
+ FIXP_SGL sfb_nrg_prev[MAX_FREQ_COEFFS]; /*!< Previous envelope (required for differential-coded values) */
+ FIXP_SGL prevNoiseLevel[MAX_NOISE_COEFFS]; /*!< Previous noise envelope (required for differential-coded values) */
+ COUPLING_MODE coupling; /*!< Stereo-mode of previous frame */
+ INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Previous strength of filtering in transposer */
+ UCHAR ampRes; /*!< Previous amplitude resolution (0: 1.5dB, 1: 3dB) */
+ UCHAR stopPos; /*!< Position in time where last envelope ended */
+ UCHAR frameErrorFlag; /*!< Previous frame status */
+}
+SBR_PREV_FRAME_DATA;
+
+typedef SBR_PREV_FRAME_DATA *HANDLE_SBR_PREV_FRAME_DATA;
+
+
+typedef struct
+{
+ int nScaleFactors; /*!< total number of scalefactors in frame */
+
+ FRAME_INFO frameInfo; /*!< time grid for current frame */
+ UCHAR domain_vec[MAX_ENVELOPES]; /*!< Bitfield containing direction of delta-coding for each envelope (0:frequency, 1:time) */
+ UCHAR domain_vec_noise[MAX_NOISE_ENVELOPES]; /*!< Same as above, but for noise envelopes */
+
+ INVF_MODE sbr_invf_mode[MAX_INVF_BANDS]; /*!< Strength of filtering in transposer */
+ COUPLING_MODE coupling; /*!< Stereo-mode */
+ int ampResolutionCurrentFrame; /*!< Amplitude resolution of envelope values (0: 1.5dB, 1: 3dB) */
+
+ UCHAR addHarmonics[MAX_FREQ_COEFFS]; /*!< Flags for synthetic sine addition */
+
+ FIXP_SGL iEnvelope[MAX_NUM_ENVELOPE_VALUES]; /*!< Envelope data */
+ FIXP_SGL sbrNoiseFloorLevel[MAX_NUM_NOISE_VALUES]; /*!< Noise envelope data */
+}
+SBR_FRAME_DATA;
+
+typedef SBR_FRAME_DATA *HANDLE_SBR_FRAME_DATA;
+
+void initSbrPrevFrameData (HANDLE_SBR_PREV_FRAME_DATA h_prev_data,
+ int timeSlots);
+
+
+int sbrGetSingleChannelElement (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameData,
+ HANDLE_FDK_BITSTREAM hBitBuf,
+ HANDLE_PS_DEC hParametricStereoDec,
+ const UINT flags,
+ const int overlap
+ );
+
+int sbrGetChannelPairElement (HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameDataLeft,
+ HANDLE_SBR_FRAME_DATA hFrameDataRight,
+ HANDLE_FDK_BITSTREAM hBitBuf,
+ const UINT flags,
+ const int overlap);
+
+SBR_HEADER_STATUS
+sbrGetHeaderData (HANDLE_SBR_HEADER_DATA headerData,
+ HANDLE_FDK_BITSTREAM hBitBuf,
+ const UINT flags,
+ const int fIsSbrData);
+
+/*!
+ \brief Initialize SBR header data
+
+ Copy default values to the header data struct and patch some entries
+ depending on the core codec.
+*/
+SBR_ERROR
+initHeaderData (
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ const int sampleRateIn,
+ const int sampleRateOut,
+ const int samplesPerFrame,
+ const UINT flags
+ );
+#endif
diff --git a/libSBRdec/src/huff_dec.cpp b/libSBRdec/src/huff_dec.cpp
new file mode 100644
index 0000000..6e00b23
--- /dev/null
+++ b/libSBRdec/src/huff_dec.cpp
@@ -0,0 +1,120 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Huffman Decoder
+*/
+
+#include "huff_dec.h"
+
+/***************************************************************************/
+/*!
+ \brief Decodes one huffman code word
+
+ Reads bits from the bitstream until a valid codeword is found.
+ The table entries are interpreted either as index to the next entry
+ or - if negative - as the codeword.
+
+ \return decoded value
+
+ \author
+
+****************************************************************************/
+int
+DecodeHuffmanCW (Huffman h, /*!< pointer to huffman codebook table */
+ HANDLE_FDK_BITSTREAM hBs) /*!< Handle to Bitbuffer */
+{
+ SCHAR index = 0;
+ int value, bit;
+
+ while (index >= 0) {
+ bit = FDKreadBits (hBs, 1);
+ index = h[index][bit];
+ }
+
+ value = index+64; /* Add offset */
+
+
+ return value;
+}
diff --git a/libSBRdec/src/huff_dec.h b/libSBRdec/src/huff_dec.h
new file mode 100644
index 0000000..7b2b50b
--- /dev/null
+++ b/libSBRdec/src/huff_dec.h
@@ -0,0 +1,100 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Huffman Decoder
+*/
+#ifndef __HUFF_DEC_H
+#define __HUFF_DEC_H
+
+#include "sbrdecoder.h"
+#include "FDK_bitstream.h"
+
+typedef const SCHAR (*Huffman)[2];
+
+int
+DecodeHuffmanCW (Huffman h,
+ HANDLE_FDK_BITSTREAM hBitBuf);
+
+#endif
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
new file mode 100644
index 0000000..2d51831
--- /dev/null
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -0,0 +1,1000 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Low Power Profile Transposer,
+ This module provides the transposer. The main entry point is lppTransposer(). The function generates
+ high frequency content by copying data from the low band (provided by core codec) into the high band.
+ This process is also referred to as "patching". The function also implements spectral whitening by means of
+ inverse filtering based on LPC coefficients.
+
+ Together with the QMF filterbank the transposer can be tested using a supplied test program. See main_audio.cpp for details.
+ This module does use fractional arithmetic and the accuracy of the computations has an impact on the overall sound quality.
+ The module also needs to take into account the different scaling of spectral data.
+
+ \sa lppTransposer(), main_audio.cpp, sbr_scale.h, \ref documentationOverview
+*/
+
+#include "lpp_tran.h"
+
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+
+#include "genericStds.h"
+#include "autocorr2nd.h"
+
+
+
+#if defined(__arm__)
+#include "arm/lpp_tran_arm.cpp"
+#endif
+
+
+
+#define LPC_SCALE_FACTOR 2
+
+
+/*!
+ *
+ * \brief Get bandwidth expansion factor from filtering level
+ *
+ * Returns a filter parameter (bandwidth expansion factor) depending on
+ * the desired filtering level signalled in the bitstream.
+ * When switching the filtering level from LOW to OFF, an additional
+ * level is being inserted to achieve a smooth transition.
+ */
+
+#ifndef FUNCTION_mapInvfMode
+static FIXP_DBL
+mapInvfMode (INVF_MODE mode,
+ INVF_MODE prevMode,
+ WHITENING_FACTORS whFactors)
+{
+ switch (mode) {
+ case INVF_LOW_LEVEL:
+ if(prevMode == INVF_OFF)
+ return whFactors.transitionLevel;
+ else
+ return whFactors.lowLevel;
+
+ case INVF_MID_LEVEL:
+ return whFactors.midLevel;
+
+ case INVF_HIGH_LEVEL:
+ return whFactors.highLevel;
+
+ default:
+ if(prevMode == INVF_LOW_LEVEL)
+ return whFactors.transitionLevel;
+ else
+ return whFactors.off;
+ }
+}
+#endif /* #ifndef FUNCTION_mapInvfMode */
+
+/*!
+ *
+ * \brief Perform inverse filtering level emphasis
+ *
+ * Retrieve bandwidth expansion factor and apply smoothing for each filter band
+ *
+ */
+
+#ifndef FUNCTION_inverseFilteringLevelEmphasis
+static void
+inverseFilteringLevelEmphasis(HANDLE_SBR_LPP_TRANS hLppTrans,/*!< Handle of lpp transposer */
+ UCHAR nInvfBands, /*!< Number of bands for inverse filtering */
+ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */
+ INVF_MODE *sbr_invf_mode_prev, /*!< Previous inverse filtering modes */
+ FIXP_DBL * bwVector /*!< Resulting filtering levels */
+ )
+{
+ for(int i = 0; i < nInvfBands; i++) {
+ FIXP_DBL accu;
+ FIXP_DBL bwTmp = mapInvfMode (sbr_invf_mode[i],
+ sbr_invf_mode_prev[i],
+ hLppTrans->pSettings->whFactors);
+
+ if(bwTmp < hLppTrans->bwVectorOld[i]) {
+ accu = fMultDiv2(FL2FXCONST_DBL(0.75f),bwTmp) +
+ fMultDiv2(FL2FXCONST_DBL(0.25f),hLppTrans->bwVectorOld[i]);
+ }
+ else {
+ accu = fMultDiv2(FL2FXCONST_DBL(0.90625f),bwTmp) +
+ fMultDiv2(FL2FXCONST_DBL(0.09375f),hLppTrans->bwVectorOld[i]);
+ }
+
+ if (accu < FL2FXCONST_DBL(0.015625f)>>1)
+ bwVector[i] = FL2FXCONST_DBL(0.0f);
+ else
+ bwVector[i] = fixMin(accu<<1,FL2FXCONST_DBL(0.99609375f));
+ }
+}
+#endif /* #ifndef FUNCTION_inverseFilteringLevelEmphasis */
+
+/* Resulting autocorrelation determinant exponent */
+#define ACDET_EXP (2*(DFRACT_BITS+sbrScaleFactor->lb_scale+10-ac.det_scale))
+#define AC_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR)
+#define ALPHA_EXP (-sbrScaleFactor->lb_scale+LPC_SCALE_FACTOR+1)
+/* Resulting transposed QMF values exponent 16 bit normalized samplebits assumed. */
+#define QMFOUT_EXP ((SAMPLE_BITS-15)-sbrScaleFactor->lb_scale)
+
+/*!
+ *
+ * \brief Perform transposition by patching of subband samples.
+ * This function serves as the main entry point into the module. The function determines the areas for the
+ * patching process (these are the source range as well as the target range) and implements spectral whitening
+ * by means of inverse filtering. The function autoCorrelation2nd() is an auxiliary function for calculating the
+ * LPC coefficients for the filtering. The actual calculation of the LPC coefficients and the implementation
+ * of the filtering are done as part of lppTransposer().
+ *
+ * Note that the filtering is done on all available QMF subsamples, whereas the patching is only done on those QMF
+ * subsamples that will be used in the next QMF synthesis. The filtering is also implemented before the patching
+ * includes further dependencies on parameters from the SBR data.
+ *
+ */
+
+void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */
+ QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */
+ FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband samples (source) */
+
+ FIXP_DBL *degreeAlias, /*!< Vector for results of aliasing estimation */
+ FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of subband samples (source) */
+ const int useLP,
+ const int timeStep, /*!< Time step of envelope */
+ const int firstSlotOffs, /*!< Start position in time */
+ const int lastSlotOffs, /*!< Number of overlap-slots into next frame */
+ const int nInvfBands, /*!< Number of bands for inverse filtering */
+ INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */
+ INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */
+ )
+{
+ INT bwIndex[MAX_NUM_PATCHES];
+ FIXP_DBL bwVector[MAX_NUM_PATCHES]; /*!< pole moving factors */
+
+ int i;
+ int loBand, start, stop;
+ TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+ PATCH_PARAM *patchParam = pSettings->patchParam;
+ int patch;
+
+ FIXP_SGL alphar[LPC_ORDER], a0r, a1r;
+ FIXP_SGL alphai[LPC_ORDER], a0i=0, a1i=0;
+ FIXP_SGL bw = FL2FXCONST_SGL(0.0f);
+
+ int autoCorrLength;
+
+ FIXP_DBL k1, k1_below=0, k1_below2=0;
+
+ ACORR_COEFS ac;
+ int startSample;
+ int stopSample;
+ int stopSampleClear;
+
+ int comLowBandScale;
+ int ovLowBandShift;
+ int lowBandShift;
+/* int ovHighBandShift;*/
+ int targetStopBand;
+
+
+ alphai[0] = FL2FXCONST_SGL(0.0f);
+ alphai[1] = FL2FXCONST_SGL(0.0f);
+
+
+ startSample = firstSlotOffs * timeStep;
+ stopSample = pSettings->nCols + lastSlotOffs * timeStep;
+
+
+ inverseFilteringLevelEmphasis(hLppTrans, nInvfBands, sbr_invf_mode, sbr_invf_mode_prev, bwVector);
+
+ stopSampleClear = stopSample;
+
+ autoCorrLength = pSettings->nCols + pSettings->overlap;
+
+ /* Set upper subbands to zero:
+ This is required in case that the patches do not cover the complete highband
+ (because the last patch would be too short).
+ Possible optimization: Clearing bands up to usb would be sufficient here. */
+ targetStopBand = patchParam[pSettings->noOfPatches-1].targetStartBand
+ + patchParam[pSettings->noOfPatches-1].numBandsInPatch;
+
+ int memSize = ((64) - targetStopBand) * sizeof(FIXP_DBL);
+
+ if (!useLP) {
+ for (i = startSample; i < stopSampleClear; i++) {
+ FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+ FDKmemclear(&qmfBufferImag[i][targetStopBand], memSize);
+ }
+ } else
+ for (i = startSample; i < stopSampleClear; i++) {
+ FDKmemclear(&qmfBufferReal[i][targetStopBand], memSize);
+ }
+
+ /* init bwIndex for each patch */
+ FDKmemclear(bwIndex, pSettings->noOfPatches*sizeof(INT));
+
+ /*
+ Calc common low band scale factor
+ */
+ comLowBandScale = fixMin(sbrScaleFactor->ov_lb_scale,sbrScaleFactor->lb_scale);
+
+ ovLowBandShift = sbrScaleFactor->ov_lb_scale - comLowBandScale;
+ lowBandShift = sbrScaleFactor->lb_scale - comLowBandScale;
+ /* ovHighBandShift = firstSlotOffs == 0 ? ovLowBandShift:0;*/
+
+ /* outer loop over bands to do analysis only once for each band */
+
+ if (!useLP) {
+ start = pSettings->lbStartPatching;
+ stop = pSettings->lbStopPatching;
+ } else
+ {
+ start = fixMax(1, pSettings->lbStartPatching - 2);
+ stop = patchParam[0].targetStartBand;
+ }
+
+
+ for ( loBand = start; loBand < stop; loBand++ ) {
+
+ FIXP_DBL lowBandReal[(((1024)/(32))+(6))+LPC_ORDER];
+ FIXP_DBL *plowBandReal = lowBandReal;
+ FIXP_DBL **pqmfBufferReal = qmfBufferReal;
+ FIXP_DBL lowBandImag[(((1024)/(32))+(6))+LPC_ORDER];
+ FIXP_DBL *plowBandImag = lowBandImag;
+ FIXP_DBL **pqmfBufferImag = qmfBufferImag;
+ int resetLPCCoeffs=0;
+ int dynamicScale = DFRACT_BITS-1-LPC_SCALE_FACTOR;
+ int acDetScale = 0; /* scaling of autocorrelation determinant */
+
+ for(i=0;i<LPC_ORDER;i++){
+ *plowBandReal++ = hLppTrans->lpcFilterStatesReal[i][loBand];
+ if (!useLP)
+ *plowBandImag++ = hLppTrans->lpcFilterStatesImag[i][loBand];
+ }
+
+ /*
+ Take old slope length qmf slot source values out of (overlap)qmf buffer
+ */
+ if (!useLP) {
+ for(i=0;i<pSettings->nCols+pSettings->overlap;i++){
+ *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+ *plowBandImag++ = (*pqmfBufferImag++)[loBand];
+ }
+ } else
+ {
+ /* pSettings->overlap is always even */
+ FDK_ASSERT((pSettings->overlap & 1) == 0);
+
+ for(i=0;i<((pSettings->overlap+pSettings->nCols)>>1);i++) {
+ *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+ *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+ }
+ if (pSettings->nCols & 1) {
+ *plowBandReal++ = (*pqmfBufferReal++)[loBand];
+ }
+ }
+
+ /*
+ Determine dynamic scaling value.
+ */
+ dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandReal, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+ dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+ if (!useLP) {
+ dynamicScale = fixMin(dynamicScale, getScalefactor(lowBandImag, LPC_ORDER+pSettings->overlap) + ovLowBandShift);
+ dynamicScale = fixMin(dynamicScale, getScalefactor(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols) + lowBandShift);
+ }
+ dynamicScale = fixMax(0, dynamicScale-1); /* one additional bit headroom to prevent -1.0 */
+
+ /*
+ Scale temporal QMF buffer.
+ */
+ scaleValues(&lowBandReal[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+ scaleValues(&lowBandReal[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+
+ if (!useLP) {
+ scaleValues(&lowBandImag[0], LPC_ORDER+pSettings->overlap, dynamicScale-ovLowBandShift);
+ scaleValues(&lowBandImag[LPC_ORDER+pSettings->overlap], pSettings->nCols, dynamicScale-lowBandShift);
+ }
+
+
+ if (!useLP) {
+ acDetScale += autoCorr2nd_cplx(&ac, lowBandReal+LPC_ORDER, lowBandImag+LPC_ORDER, autoCorrLength);
+ }
+ else
+ {
+ acDetScale += autoCorr2nd_real(&ac, lowBandReal+LPC_ORDER, autoCorrLength);
+ }
+
+ /* Examine dynamic of determinant in autocorrelation. */
+ acDetScale += 2*(comLowBandScale + dynamicScale);
+ acDetScale *= 2; /* two times reflection coefficent scaling */
+ acDetScale += ac.det_scale; /* ac scaling of determinant */
+
+ /* In case of determinant < 10^-38, resetLPCCoeffs=1 has to be enforced. */
+ if (acDetScale>126 ) {
+ resetLPCCoeffs = 1;
+ }
+
+
+ alphar[1] = FL2FXCONST_SGL(0.0f);
+ if (!useLP)
+ alphai[1] = FL2FXCONST_SGL(0.0f);
+
+ if (ac.det != FL2FXCONST_DBL(0.0f)) {
+ FIXP_DBL tmp,absTmp,absDet;
+
+ absDet = fixp_abs(ac.det);
+
+ if (!useLP) {
+ tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+ ( (fMultDiv2(ac.r01i,ac.r12i) + fMultDiv2(ac.r02r,ac.r11r)) >> (LPC_SCALE_FACTOR-1) );
+ } else
+ {
+ tmp = ( fMultDiv2(ac.r01r,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) -
+ ( fMultDiv2(ac.r02r,ac.r11r) >> (LPC_SCALE_FACTOR-1) );
+ }
+ absTmp = fixp_abs(tmp);
+
+ /*
+ Quick check: is first filter coeff >= 1(4)
+ */
+ {
+ INT scale;
+ FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+ scale = scale+ac.det_scale;
+
+ if ( (scale > 0) && (result >= (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+ resetLPCCoeffs = 1;
+ }
+ else {
+ alphar[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+ if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+ alphar[1] = -alphar[1];
+ }
+ }
+ }
+
+ if (!useLP)
+ {
+ tmp = ( fMultDiv2(ac.r01i,ac.r12r) >> (LPC_SCALE_FACTOR-1) ) +
+ ( (fMultDiv2(ac.r01r,ac.r12i) - (FIXP_DBL)fMultDiv2(ac.r02i,ac.r11r)) >> (LPC_SCALE_FACTOR-1) ) ;
+
+ absTmp = fixp_abs(tmp);
+
+ /*
+ Quick check: is second filter coeff >= 1(4)
+ */
+ {
+ INT scale;
+ FIXP_DBL result = fDivNorm(absTmp, absDet, &scale);
+ scale = scale+ac.det_scale;
+
+ if ( (scale > 0) && (result >= /*FL2FXCONST_DBL(1.f)*/ (FIXP_DBL)MAXVAL_DBL>>scale) ) {
+ resetLPCCoeffs = 1;
+ }
+ else {
+ alphai[1] = FX_DBL2FX_SGL(scaleValue(result,scale));
+ if((tmp<FL2FX_DBL(0.0f)) ^ (ac.det<FL2FX_DBL(0.0f))) {
+ alphai[1] = -alphai[1];
+ }
+ }
+ }
+ }
+ }
+
+ alphar[0] = FL2FXCONST_SGL(0.0f);
+ if (!useLP)
+ alphai[0] = FL2FXCONST_SGL(0.0f);
+
+ if ( ac.r11r != FL2FXCONST_DBL(0.0f) ) {
+
+ /* ac.r11r is always >=0 */
+ FIXP_DBL tmp,absTmp;
+
+ if (!useLP) {
+ tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) +
+ (fMultDiv2(alphar[1],ac.r12r) + fMultDiv2(alphai[1],ac.r12i));
+ } else
+ {
+ if(ac.r01r>=FL2FXCONST_DBL(0.0f))
+ tmp = (ac.r01r>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+ else
+ tmp = -((-ac.r01r)>>(LPC_SCALE_FACTOR+1)) + fMultDiv2(alphar[1],ac.r12r);
+ }
+
+ absTmp = fixp_abs(tmp);
+
+ /*
+ Quick check: is first filter coeff >= 1(4)
+ */
+
+ if (absTmp >= (ac.r11r>>1)) {
+ resetLPCCoeffs=1;
+ }
+ else {
+ INT scale;
+ FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+ alphar[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1));
+
+ if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+ alphar[0] = -alphar[0];
+ }
+
+ if (!useLP)
+ {
+ tmp = (ac.r01i>>(LPC_SCALE_FACTOR+1)) +
+ (fMultDiv2(alphai[1],ac.r12r) - fMultDiv2(alphar[1],ac.r12i));
+
+ absTmp = fixp_abs(tmp);
+
+ /*
+ Quick check: is second filter coeff >= 1(4)
+ */
+ if (absTmp >= (ac.r11r>>1)) {
+ resetLPCCoeffs=1;
+ }
+ else {
+ INT scale;
+ FIXP_DBL result = fDivNorm(absTmp, fixp_abs(ac.r11r), &scale);
+ alphai[0] = FX_DBL2FX_SGL(scaleValue(result,scale+1));
+ if((tmp>FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))
+ alphai[0] = -alphai[0];
+ }
+ }
+ }
+
+
+ if (!useLP)
+ {
+ /* Now check the quadratic criteria */
+ if( (fMultDiv2(alphar[0],alphar[0]) + fMultDiv2(alphai[0],alphai[0])) >= FL2FXCONST_DBL(0.5f) )
+ resetLPCCoeffs=1;
+ if( (fMultDiv2(alphar[1],alphar[1]) + fMultDiv2(alphai[1],alphai[1])) >= FL2FXCONST_DBL(0.5f) )
+ resetLPCCoeffs=1;
+ }
+
+ if(resetLPCCoeffs){
+ alphar[0] = FL2FXCONST_SGL(0.0f);
+ alphar[1] = FL2FXCONST_SGL(0.0f);
+ if (!useLP)
+ {
+ alphai[0] = FL2FXCONST_SGL(0.0f);
+ alphai[1] = FL2FXCONST_SGL(0.0f);
+ }
+ }
+
+ if (useLP)
+ {
+
+ /* Aliasing detection */
+ if(ac.r11r==FL2FXCONST_DBL(0.0f)) {
+ k1 = FL2FXCONST_DBL(0.0f);
+ }
+ else {
+ if ( fixp_abs(ac.r01r) >= fixp_abs(ac.r11r) ) {
+ if ( fMultDiv2(ac.r01r,ac.r11r) < FL2FX_DBL(0.0f)) {
+ k1 = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_SGL(1.0f)*/;
+ }else {
+ /* Since this value is squared later, it must not ever become -1.0f. */
+ k1 = (FIXP_DBL)(MINVAL_DBL+1) /*FL2FXCONST_SGL(-1.0f)*/;
+ }
+ }
+ else {
+ INT scale;
+ FIXP_DBL result = fDivNorm(fixp_abs(ac.r01r), fixp_abs(ac.r11r), &scale);
+ k1 = scaleValue(result,scale);
+
+ if(!((ac.r01r<FL2FX_DBL(0.0f)) ^ (ac.r11r<FL2FX_DBL(0.0f)))) {
+ k1 = -k1;
+ }
+ }
+ }
+ if(loBand > 1){
+ /* Check if the gain should be locked */
+ FIXP_DBL deg = /*FL2FXCONST_DBL(1.0f)*/ (FIXP_DBL)MAXVAL_DBL - fPow2(k1_below);
+ degreeAlias[loBand] = FL2FXCONST_DBL(0.0f);
+ if (((loBand & 1) == 0) && (k1 < FL2FXCONST_DBL(0.0f))){
+ if (k1_below < FL2FXCONST_DBL(0.0f)) { /* 2-Ch Aliasing Detection */
+ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+ if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */
+ degreeAlias[loBand-1] = deg;
+ }
+ }
+ else if ( k1_below2 > FL2FXCONST_DBL(0.0f) ) { /* 3-Ch Aliasing Detection */
+ degreeAlias[loBand] = deg;
+ }
+ }
+ if (((loBand & 1) == 1) && (k1 > FL2FXCONST_DBL(0.0f))){
+ if (k1_below > FL2FXCONST_DBL(0.0f)) { /* 2-CH Aliasing Detection */
+ degreeAlias[loBand] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+ if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */
+ degreeAlias[loBand-1] = deg;
+ }
+ }
+ else if ( k1_below2 < FL2FXCONST_DBL(0.0f) ) { /* 3-CH Aliasing Detection */
+ degreeAlias[loBand] = deg;
+ }
+ }
+ }
+ /* remember k1 values of the 2 QMF channels below the current channel */
+ k1_below2 = k1_below;
+ k1_below = k1;
+ }
+
+ patch = 0;
+
+ while ( patch < pSettings->noOfPatches ) { /* inner loop over every patch */
+
+ int hiBand = loBand + patchParam[patch].targetBandOffs;
+
+ if ( loBand < patchParam[patch].sourceStartBand
+ || loBand >= patchParam[patch].sourceStopBand
+ //|| hiBand >= hLppTrans->pSettings->noChannels
+ ) {
+ /* Lowband not in current patch - proceed */
+ patch++;
+ continue;
+ }
+
+ FDK_ASSERT( hiBand < (64) );
+
+ /* bwIndex[patch] is already initialized with value from previous band inside this patch */
+ while (hiBand >= pSettings->bwBorders[bwIndex[patch]])
+ bwIndex[patch]++;
+
+
+ /*
+ Filter Step 2: add the left slope with the current filter to the buffer
+ pure source values are already in there
+ */
+ bw = FX_DBL2FX_SGL(bwVector[bwIndex[patch]]);
+
+ a0r = FX_DBL2FX_SGL(fMult(bw,alphar[0])); /* Apply current bandwidth expansion factor */
+
+
+ if (!useLP)
+ a0i = FX_DBL2FX_SGL(fMult(bw,alphai[0]));
+ bw = FX_DBL2FX_SGL(fPow2(bw));
+ a1r = FX_DBL2FX_SGL(fMult(bw,alphar[1]));
+ if (!useLP)
+ a1i = FX_DBL2FX_SGL(fMult(bw,alphai[1]));
+
+
+
+ /*
+ Filter Step 3: insert the middle part which won't be windowed
+ */
+
+ if ( bw <= FL2FXCONST_SGL(0.0f) ) {
+ if (!useLP) {
+ int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+ for(i = startSample; i < stopSample; i++ ) {
+ qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+ qmfBufferImag[i][hiBand] = lowBandImag[LPC_ORDER+i]>>descale;
+ }
+ } else
+ {
+ int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+ for(i = startSample; i < stopSample; i++ ) {
+ qmfBufferReal[i][hiBand] = lowBandReal[LPC_ORDER+i]>>descale;
+ }
+ }
+ }
+ else { /* bw <= 0 */
+
+ if (!useLP) {
+ int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+#ifdef FUNCTION_LPPTRANSPOSER_func1
+ lppTransposer_func1(lowBandReal+LPC_ORDER+startSample,lowBandImag+LPC_ORDER+startSample,
+ qmfBufferReal+startSample,qmfBufferImag+startSample,
+ stopSample-startSample, (int) hiBand,
+ dynamicScale,descale,
+ a0r, a0i, a1r, a1i);
+#else
+ for(i = startSample; i < stopSample; i++ ) {
+ FIXP_DBL accu1, accu2;
+
+ accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) - fMultDiv2(a0i,lowBandImag[LPC_ORDER+i-1]) +
+ fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]) - fMultDiv2(a1i,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+ accu2 = (fMultDiv2(a0i,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a0r,lowBandImag[LPC_ORDER+i-1]) +
+ fMultDiv2(a1i,lowBandReal[LPC_ORDER+i-2]) + fMultDiv2(a1r,lowBandImag[LPC_ORDER+i-2]))>>dynamicScale;
+
+ qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+ qmfBufferImag[i][hiBand] = (lowBandImag[LPC_ORDER+i]>>descale) + (accu2<<1);
+ }
+#endif
+ } else
+ {
+ int descale = fixMin(DFRACT_BITS-1, (LPC_SCALE_FACTOR+dynamicScale));
+
+ FDK_ASSERT(dynamicScale >= 0);
+ for(i = startSample; i < stopSample; i++ ) {
+ FIXP_DBL accu1;
+
+ accu1 = (fMultDiv2(a0r,lowBandReal[LPC_ORDER+i-1]) + fMultDiv2(a1r,lowBandReal[LPC_ORDER+i-2]))>>dynamicScale;
+
+ qmfBufferReal[i][hiBand] = (lowBandReal[LPC_ORDER+i]>>descale) + (accu1<<1);
+ }
+ }
+ } /* bw <= 0 */
+
+ patch++;
+
+ } /* inner loop over patches */
+
+ /*
+ * store the unmodified filter coefficients if there is
+ * an overlapping envelope
+ *****************************************************************/
+
+
+ } /* outer loop over bands (loBand) */
+
+ if (useLP)
+ {
+ for ( loBand = pSettings->lbStartPatching; loBand < pSettings->lbStopPatching; loBand++ ) {
+ patch = 0;
+ while ( patch < pSettings->noOfPatches ) {
+
+ UCHAR hiBand = loBand + patchParam[patch].targetBandOffs;
+
+ if ( loBand < patchParam[patch].sourceStartBand
+ || loBand >= patchParam[patch].sourceStopBand
+ || hiBand >= (64) /* Highband out of range (biterror) */
+ ) {
+ /* Lowband not in current patch or highband out of range (might be caused by biterrors)- proceed */
+ patch++;
+ continue;
+ }
+
+ if(hiBand != patchParam[patch].targetStartBand)
+ degreeAlias[hiBand] = degreeAlias[loBand];
+
+ patch++;
+ }
+ }/* end for loop */
+ }
+
+ for (i = 0; i < nInvfBands; i++ ) {
+ hLppTrans->bwVectorOld[i] = bwVector[i];
+ }
+
+ /*
+ set high band scale factor
+ */
+ sbrScaleFactor->hb_scale = comLowBandScale-(LPC_SCALE_FACTOR);
+
+}
+
+/*!
+ *
+ * \brief Initialize one low power transposer instance
+ *
+ *
+ */
+SBR_ERROR
+createLppTransposer (HANDLE_SBR_LPP_TRANS hs, /*!< Handle of low power transposer */
+ TRANSPOSER_SETTINGS *pSettings, /*!< Pointer to settings */
+ const int highBandStartSb, /*!< ? */
+ UCHAR *v_k_master, /*!< Master table */
+ const int numMaster, /*!< Valid entries in master table */
+ const int usb, /*!< Highband area stop subband */
+ const int timeSlots, /*!< Number of time slots */
+ const int nCols, /*!< Number of colums (codec qmf bank) */
+ UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */
+ const int noNoiseBands, /*!< Number of noise bands */
+ UINT fs, /*!< Sample Frequency */
+ const int chan, /*!< Channel number */
+ const int overlap
+ )
+{
+ /* FB inverse filtering settings */
+ hs->pSettings = pSettings;
+
+ pSettings->nCols = nCols;
+ pSettings->overlap = overlap;
+
+ switch (timeSlots) {
+
+ case 15:
+ case 16:
+ break;
+
+ default:
+ return SBRDEC_UNSUPPORTED_CONFIG; /* Unimplemented */
+ }
+
+ if (chan==0) {
+ /* Init common data only once */
+ hs->pSettings->nCols = nCols;
+
+ return resetLppTransposer (hs,
+ highBandStartSb,
+ v_k_master,
+ numMaster,
+ noiseBandTable,
+ noNoiseBands,
+ usb,
+ fs);
+ }
+ return SBRDEC_OK;
+}
+
+
+static int findClosestEntry(UCHAR goalSb, UCHAR *v_k_master, UCHAR numMaster, UCHAR direction)
+{
+ int index;
+
+ if( goalSb <= v_k_master[0] )
+ return v_k_master[0];
+
+ if( goalSb >= v_k_master[numMaster] )
+ return v_k_master[numMaster];
+
+ if(direction) {
+ index = 0;
+ while( v_k_master[index] < goalSb ) {
+ index++;
+ }
+ } else {
+ index = numMaster;
+ while( v_k_master[index] > goalSb ) {
+ index--;
+ }
+ }
+
+ return v_k_master[index];
+}
+
+
+/*!
+ *
+ * \brief Reset memory for one lpp transposer instance
+ *
+ * \return SBRDEC_OK on success, SBRDEC_UNSUPPORTED_CONFIG on error
+ */
+SBR_ERROR
+resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */
+ UCHAR highBandStartSb, /*!< High band area: start subband */
+ UCHAR *v_k_master, /*!< Master table */
+ UCHAR numMaster, /*!< Valid entries in master table */
+ UCHAR *noiseBandTable, /*!< Mapping of SBR noise bands to QMF bands */
+ UCHAR noNoiseBands, /*!< Number of noise bands */
+ UCHAR usb, /*!< High band area: stop subband */
+ UINT fs /*!< SBR output sampling frequency */
+ )
+{
+ TRANSPOSER_SETTINGS *pSettings = hLppTrans->pSettings;
+ PATCH_PARAM *patchParam = pSettings->patchParam;
+
+ int i, patch;
+ int targetStopBand;
+ int sourceStartBand;
+ int patchDistance;
+ int numBandsInPatch;
+
+ int lsb = v_k_master[0]; /* Start subband expressed in "non-critical" sampling terms*/
+ int xoverOffset = highBandStartSb - lsb; /* Calculate distance in QMF bands between k0 and kx */
+ int startFreqHz;
+
+ int desiredBorder;
+
+ usb = fixMin(usb, v_k_master[numMaster]); /* Avoid endless loops (compare with float code). */
+
+ /*
+ * Plausibility check
+ */
+
+ if ( lsb - SHIFT_START_SB < 4 ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+
+ /*
+ * Initialize the patching parameter
+ */
+ desiredBorder = 21;
+ if (fs < 92017) {
+ desiredBorder = 23;
+ }
+ if (fs < 75132) {
+ desiredBorder = 32;
+ }
+ if (fs < 55426) {
+ desiredBorder = 43;
+ }
+ if (fs < 46009) {
+ desiredBorder = 46;
+ }
+ if (fs < 35777) {
+ desiredBorder = 64;
+ }
+
+ desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */
+
+ /* First patch */
+ sourceStartBand = SHIFT_START_SB + xoverOffset;
+ targetStopBand = lsb + xoverOffset; /* upperBand */
+
+ /* Even (odd) numbered channel must be patched to even (odd) numbered channel */
+ patch = 0;
+ while(targetStopBand < usb) {
+
+ /* Too many patches?
+ Allow MAX_NUM_PATCHES+1 patches here.
+ we need to check later again, since patch might be the highest patch
+ AND contain less than 3 bands => actual number of patches will be reduced by 1.
+ */
+ if (patch > MAX_NUM_PATCHES) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ patchParam[patch].guardStartBand = targetStopBand;
+ patchParam[patch].targetStartBand = targetStopBand;
+
+ numBandsInPatch = desiredBorder - targetStopBand; /* Get the desired range of the patch */
+
+ if ( numBandsInPatch >= lsb - sourceStartBand ) {
+ /* Desired number bands are not available -> patch whole source range */
+ patchDistance = targetStopBand - sourceStartBand; /* Get the targetOffset */
+ patchDistance = patchDistance & ~1; /* Rounding off odd numbers and make all even */
+ numBandsInPatch = lsb - (targetStopBand - patchDistance); /* Update number of bands to be patched */
+ numBandsInPatch = findClosestEntry(targetStopBand + numBandsInPatch, v_k_master, numMaster, 0) -
+ targetStopBand; /* Adapt region to master-table */
+ }
+
+ /* Desired number bands are available -> get the minimal even patching distance */
+ patchDistance = numBandsInPatch + targetStopBand - lsb; /* Get minimal distance */
+ patchDistance = (patchDistance + 1) & ~1; /* Rounding up odd numbers and make all even */
+
+ if (numBandsInPatch > 0) {
+ patchParam[patch].sourceStartBand = targetStopBand - patchDistance;
+ patchParam[patch].targetBandOffs = patchDistance;
+ patchParam[patch].numBandsInPatch = numBandsInPatch;
+ patchParam[patch].sourceStopBand = patchParam[patch].sourceStartBand + numBandsInPatch;
+
+ targetStopBand += patchParam[patch].numBandsInPatch;
+ patch++;
+ }
+
+ /* All patches but first */
+ sourceStartBand = SHIFT_START_SB;
+
+ /* Check if we are close to desiredBorder */
+ if( desiredBorder - targetStopBand < 3) /* MPEG doc */
+ {
+ desiredBorder = usb;
+ }
+
+ }
+
+ patch--;
+
+ /* If highest patch contains less than three subband: skip it */
+ if ( (patch>0) && (patchParam[patch].numBandsInPatch < 3) ) {
+ patch--;
+ targetStopBand = patchParam[patch].targetStartBand + patchParam[patch].numBandsInPatch;
+ }
+
+ /* now check if we don't have one too many */
+ if (patch >= MAX_NUM_PATCHES) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ pSettings->noOfPatches = patch + 1;
+
+ /* Check lowest and highest source subband */
+ pSettings->lbStartPatching = targetStopBand;
+ pSettings->lbStopPatching = 0;
+ for ( patch = 0; patch < pSettings->noOfPatches; patch++ ) {
+ pSettings->lbStartPatching = fixMin( pSettings->lbStartPatching, patchParam[patch].sourceStartBand );
+ pSettings->lbStopPatching = fixMax( pSettings->lbStopPatching, patchParam[patch].sourceStopBand );
+ }
+
+ for(i = 0 ; i < noNoiseBands; i++){
+ pSettings->bwBorders[i] = noiseBandTable[i+1];
+ }
+
+ /*
+ * Choose whitening factors
+ */
+
+ startFreqHz = ( (lsb + xoverOffset)*fs ) >> 7; /* Shift does a division by 2*(64) */
+
+ for( i = 1; i < NUM_WHFACTOR_TABLE_ENTRIES; i++ )
+ {
+ if( startFreqHz < FDK_sbrDecoder_sbr_whFactorsIndex[i])
+ break;
+ }
+ i--;
+
+ pSettings->whFactors.off = FDK_sbrDecoder_sbr_whFactorsTable[i][0];
+ pSettings->whFactors.transitionLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][1];
+ pSettings->whFactors.lowLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][2];
+ pSettings->whFactors.midLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][3];
+ pSettings->whFactors.highLevel = FDK_sbrDecoder_sbr_whFactorsTable[i][4];
+
+ return SBRDEC_OK;
+}
diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h
new file mode 100644
index 0000000..1838c06
--- /dev/null
+++ b/libSBRdec/src/lpp_tran.h
@@ -0,0 +1,242 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Low Power Profile Transposer,
+*/
+
+#ifndef _LPP_TRANS_H
+#define _LPP_TRANS_H
+
+#include "sbrdecoder.h"
+#include "qmf.h"
+
+/*
+ Common
+*/
+#define QMF_OUT_SCALE 8
+
+/*
+ Env-Adjust
+*/
+#define MAX_NOISE_ENVELOPES 2
+#define MAX_NOISE_COEFFS 5
+#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS)
+#define MAX_NUM_LIMITERS 12
+
+/* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs
+ by overriding MAX_ENVELOPES in the correct order: */
+#define MAX_ENVELOPES_HEAAC 5
+#define MAX_ENVELOPES MAX_ENVELOPES_HEAAC
+
+#define MAX_FREQ_COEFFS 48
+#define MAX_FREQ_COEFFS_FS44100 35
+#define MAX_FREQ_COEFFS_FS48000 32
+
+
+#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS)
+
+#define MAX_GAIN_EXP 34
+/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP)
+ example: 34=99dB */
+#define MAX_GAIN_CONCEAL_EXP 1
+/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case (0dB) */
+
+/*
+ LPP Transposer
+*/
+#define LPC_ORDER 2
+
+#define MAX_INVF_BANDS MAX_NOISE_COEFFS
+
+#define MAX_NUM_PATCHES 6
+#define SHIFT_START_SB 1 /*!< lowest subband of source range */
+
+typedef enum
+{
+ INVF_OFF = 0,
+ INVF_LOW_LEVEL,
+ INVF_MID_LEVEL,
+ INVF_HIGH_LEVEL,
+ INVF_SWITCHED /* not a real choice but used here to control behaviour */
+}
+INVF_MODE;
+
+
+/** parameter set for one single patch */
+typedef struct {
+ UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples from */
+ UCHAR sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */
+ UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in order to
+ reduce interferences between patches */
+ UCHAR targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */
+ UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */
+ UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */
+} PATCH_PARAM;
+
+
+/** whitening factors for different levels of whitening
+ need to be initialized corresponding to crossover frequency */
+typedef struct {
+ FIXP_DBL off; /*!< bw factor for signal OFF */
+ FIXP_DBL transitionLevel;
+ FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */
+ FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */
+ FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */
+} WHITENING_FACTORS;
+
+
+/*! The transposer settings are calculated on a header reset and are shared by both channels. */
+typedef struct {
+ UCHAR nCols; /*!< number subsamples of a codec frame */
+ UCHAR noOfPatches; /*!< number of patches */
+ UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */
+ UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/
+ UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different inverse filtering levels */
+
+ PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */
+ WHITENING_FACTORS whFactors; /*!< the pole moving factors for certain whitening levels as indicated
+ in the bitstream depending on the crossover frequency */
+ UCHAR overlap; /*!< Overlap size */
+} TRANSPOSER_SETTINGS;
+
+
+typedef struct
+{
+ TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */
+ FIXP_DBL bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */
+ FIXP_DBL lpcFilterStatesReal[LPC_ORDER][(32)]; /*!< pointer array to save filter states */
+ FIXP_DBL lpcFilterStatesImag[LPC_ORDER][(32)]; /*!< pointer array to save filter states */
+}
+SBR_LPP_TRANS;
+
+typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS;
+
+
+void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+ QMF_SCALE_FACTOR *sbrScaleFactor,
+ FIXP_DBL **qmfBufferReal,
+
+ FIXP_DBL *degreeAlias,
+ FIXP_DBL **qmfBufferImag,
+ const int useLP,
+ const int timeStep,
+ const int firstSlotOffset,
+ const int lastSlotOffset,
+ const int nInvfBands,
+ INVF_MODE *sbr_invf_mode,
+ INVF_MODE *sbr_invf_mode_prev
+ );
+
+
+SBR_ERROR
+createLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+ TRANSPOSER_SETTINGS *pSettings,
+ const int highBandStartSb,
+ UCHAR *v_k_master,
+ const int numMaster,
+ const int usb,
+ const int timeSlots,
+ const int nCols,
+ UCHAR *noiseBandTable,
+ const int noNoiseBands,
+ UINT fs,
+ const int chan,
+ const int overlap);
+
+
+SBR_ERROR
+resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,
+ UCHAR highBandStartSb,
+ UCHAR *v_k_master,
+ UCHAR numMaster,
+ UCHAR *noiseBandTable,
+ UCHAR noNoiseBands,
+ UCHAR usb,
+ UINT fs);
+
+
+
+#endif /* _LPP_TRANS_H */
+
diff --git a/libSBRdec/src/psbitdec.cpp b/libSBRdec/src/psbitdec.cpp
new file mode 100644
index 0000000..dfd532f
--- /dev/null
+++ b/libSBRdec/src/psbitdec.cpp
@@ -0,0 +1,593 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+#include "psbitdec.h"
+
+
+#include "sbr_rom.h"
+#include "huff_dec.h"
+
+/* PS dec privat functions */
+SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d);
+void ResetPsDeCor (HANDLE_PS_DEC h_ps_d);
+
+/***************************************************************************/
+/*!
+ \brief huffman decoding by codebook table
+
+ \return index of huffman codebook table
+
+****************************************************************************/
+static SCHAR
+decode_huff_cw (Huffman h, /*!< pointer to huffman codebook table */
+ HANDLE_FDK_BITSTREAM hBitBuf, /*!< Handle to Bitbuffer */
+ int *length) /*!< length of huffman codeword (or NULL) */
+{
+ UCHAR bit = 0;
+ SCHAR index = 0;
+ UCHAR bitCount = 0;
+
+ while (index >= 0) {
+ bit = FDKreadBits (hBitBuf, 1);
+ bitCount++;
+ index = h[index][bit];
+ }
+ if (length) {
+ *length = bitCount;
+ }
+ return( index+64 ); /* Add offset */
+}
+
+/***************************************************************************/
+/*!
+ \brief helper function - limiting of value to min/max values
+
+ \return limited value
+
+****************************************************************************/
+
+static SCHAR
+limitMinMax(SCHAR i,
+ SCHAR min,
+ SCHAR max)
+{
+ if (i<min)
+ return min;
+ else if (i>max)
+ return max;
+ else
+ return i;
+}
+
+/***************************************************************************/
+/*!
+ \brief Decodes delta values in-place and updates
+ data buffers according to quantization classes.
+
+ When delta coded in frequency the first element is deltacode from zero.
+ aIndex buffer is decoded from delta values to actual values.
+
+ \return none
+
+****************************************************************************/
+static void
+deltaDecodeArray(SCHAR enable,
+ SCHAR *aIndex, /*!< ICC/IID parameters */
+ SCHAR *aPrevFrameIndex, /*!< ICC/IID parameters of previous frame */
+ SCHAR DtDf,
+ UCHAR nrElements, /*!< as conveyed in bitstream */
+ /*!< output array size: nrElements*stride */
+ UCHAR stride, /*!< 1=dflt, 2=half freq. resolution */
+ SCHAR minIdx,
+ SCHAR maxIdx)
+{
+ int i;
+
+ /* Delta decode */
+ if ( enable==1 ) {
+ if (DtDf == 0) { /* Delta coded in freq */
+ aIndex[0] = 0 + aIndex[0];
+ aIndex[0] = limitMinMax(aIndex[0],minIdx,maxIdx);
+ for (i = 1; i < nrElements; i++) {
+ aIndex[i] = aIndex[i-1] + aIndex[i];
+ aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx);
+ }
+ }
+ else { /* Delta time */
+ for (i = 0; i < nrElements; i++) {
+ aIndex[i] = aPrevFrameIndex[i*stride] + aIndex[i];
+ aIndex[i] = limitMinMax(aIndex[i],minIdx,maxIdx);
+ }
+ }
+ }
+ else { /* No data is sent, set index to zero */
+ for (i = 0; i < nrElements; i++) {
+ aIndex[i] = 0;
+ }
+ }
+ if (stride==2) {
+ for (i=nrElements*stride-1; i>0; i--) {
+ aIndex[i] = aIndex[i>>1];
+ }
+ }
+}
+
+/***************************************************************************/
+/*!
+ \brief Mapping of ICC/IID parameters to 20 stereo bands
+
+ \return none
+
+****************************************************************************/
+static void map34IndexTo20 (SCHAR *aIndex, /*!< decoded ICC/IID parameters */
+ UCHAR noBins) /*!< number of stereo bands */
+{
+ aIndex[0] = (2*aIndex[0]+aIndex[1])/3;
+ aIndex[1] = (aIndex[1]+2*aIndex[2])/3;
+ aIndex[2] = (2*aIndex[3]+aIndex[4])/3;
+ aIndex[3] = (aIndex[4]+2*aIndex[5])/3;
+ aIndex[4] = (aIndex[6]+aIndex[7])/2;
+ aIndex[5] = (aIndex[8]+aIndex[9])/2;
+ aIndex[6] = aIndex[10];
+ aIndex[7] = aIndex[11];
+ aIndex[8] = (aIndex[12]+aIndex[13])/2;
+ aIndex[9] = (aIndex[14]+aIndex[15])/2;
+ aIndex[10] = aIndex[16];
+ /* For IPD/OPD it stops here */
+
+ if (noBins == NO_HI_RES_BINS)
+ {
+ aIndex[11] = aIndex[17];
+ aIndex[12] = aIndex[18];
+ aIndex[13] = aIndex[19];
+ aIndex[14] = (aIndex[20]+aIndex[21])/2;
+ aIndex[15] = (aIndex[22]+aIndex[23])/2;
+ aIndex[16] = (aIndex[24]+aIndex[25])/2;
+ aIndex[17] = (aIndex[26]+aIndex[27])/2;
+ aIndex[18] = (aIndex[28]+aIndex[29]+aIndex[30]+aIndex[31])/4;
+ aIndex[19] = (aIndex[32]+aIndex[33])/2;
+ }
+}
+
+/***************************************************************************/
+/*!
+ \brief Decodes delta coded IID, ICC, IPD and OPD indices
+
+ \return PS processing flag. If set to 1
+
+****************************************************************************/
+int
+DecodePs( struct PS_DEC *h_ps_d, /*!< PS handle */
+ const UCHAR frameError ) /*!< Flag telling that frame had errors */
+{
+ MPEG_PS_BS_DATA *pBsData;
+ UCHAR gr, env;
+ int bPsHeaderValid, bPsDataAvail;
+
+ /* Shortcuts to avoid deferencing and keep the code readable */
+ pBsData = &h_ps_d->bsData[h_ps_d->processSlot].mpeg;
+ bPsHeaderValid = pBsData->bPsHeaderValid;
+ bPsDataAvail = (h_ps_d->bPsDataAvail[h_ps_d->processSlot] == ppt_mpeg) ? 1 : 0;
+
+ /***************************************************************************************
+ * Decide whether to process or to conceal PS data or not. */
+
+ if ( ( h_ps_d->psDecodedPrv && !frameError && !bPsDataAvail)
+ || (!h_ps_d->psDecodedPrv && (frameError || !bPsDataAvail || !bPsHeaderValid)) ) {
+ /* Don't apply PS processing.
+ * Declare current PS header and bitstream data invalid. */
+ pBsData->bPsHeaderValid = 0;
+ h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none;
+ return (0);
+ }
+
+ if (frameError || !bPsHeaderValid)
+ { /* no new PS data available (e.g. frame loss) */
+ /* => keep latest data constant (i.e. FIX with noEnv=0) */
+ pBsData->noEnv = 0;
+ }
+
+ /***************************************************************************************
+ * Decode bitstream payload or prepare parameter for concealment:
+ */
+ for (env=0; env<pBsData->noEnv; env++) {
+ SCHAR *aPrevIidIndex;
+ SCHAR *aPrevIccIndex;
+
+ UCHAR noIidSteps = pBsData->bFineIidQ?NO_IID_STEPS_FINE:NO_IID_STEPS;
+
+ if (env==0) {
+ aPrevIidIndex = h_ps_d->specificTo.mpeg.aIidPrevFrameIndex;
+ aPrevIccIndex = h_ps_d->specificTo.mpeg.aIccPrevFrameIndex;
+ }
+ else {
+ aPrevIidIndex = pBsData->aaIidIndex[env-1];
+ aPrevIccIndex = pBsData->aaIccIndex[env-1];
+ }
+
+ deltaDecodeArray(pBsData->bEnableIid,
+ pBsData->aaIidIndex[env],
+ aPrevIidIndex,
+ pBsData->abIidDtFlag[env],
+ FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid],
+ (pBsData->freqResIid)?1:2,
+ -noIidSteps,
+ noIidSteps);
+
+ deltaDecodeArray(pBsData->bEnableIcc,
+ pBsData->aaIccIndex[env],
+ aPrevIccIndex,
+ pBsData->abIccDtFlag[env],
+ FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc],
+ (pBsData->freqResIcc)?1:2,
+ 0,
+ NO_ICC_STEPS-1);
+ } /* for (env=0; env<pBsData->noEnv; env++) */
+
+ /* handling of FIX noEnv=0 */
+ if (pBsData->noEnv==0) {
+ /* set noEnv=1, keep last parameters or force 0 if not enabled */
+ pBsData->noEnv = 1;
+
+ if (pBsData->bEnableIid) {
+ for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+ pBsData->aaIidIndex[pBsData->noEnv-1][gr] =
+ h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr];
+ }
+ }
+ else {
+ for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+ pBsData->aaIidIndex[pBsData->noEnv-1][gr] = 0;
+ }
+ }
+
+ if (pBsData->bEnableIcc) {
+ for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+ pBsData->aaIccIndex[pBsData->noEnv-1][gr] =
+ h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr];
+ }
+ }
+ else {
+ for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+ pBsData->aaIccIndex[pBsData->noEnv-1][gr] = 0;
+ }
+ }
+ }
+
+ /* Update previous frame index buffers */
+ for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+ h_ps_d->specificTo.mpeg.aIidPrevFrameIndex[gr] =
+ pBsData->aaIidIndex[pBsData->noEnv-1][gr];
+ }
+ for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+ h_ps_d->specificTo.mpeg.aIccPrevFrameIndex[gr] =
+ pBsData->aaIccIndex[pBsData->noEnv-1][gr];
+ }
+
+ /* PS data from bitstream (if avail) was decoded now */
+ h_ps_d->bPsDataAvail[h_ps_d->processSlot] = ppt_none;
+
+ /* handling of env borders for FIX & VAR */
+ if (pBsData->bFrameClass == 0) {
+ /* FIX_BORDERS NoEnv=0,1,2,4 */
+ pBsData->aEnvStartStop[0] = 0;
+ for (env=1; env<pBsData->noEnv; env++) {
+ pBsData->aEnvStartStop[env] =
+ (env * h_ps_d->noSubSamples) / pBsData->noEnv;
+ }
+ pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples;
+ /* 1024 (32 slots) env borders: 0, 8, 16, 24, 32 */
+ /* 960 (30 slots) env borders: 0, 7, 15, 22, 30 */
+ }
+ else { /* if (h_ps_d->bFrameClass == 0) */
+ /* VAR_BORDERS NoEnv=1,2,3,4 */
+ pBsData->aEnvStartStop[0] = 0;
+
+ /* handle case aEnvStartStop[noEnv]<noSubSample for VAR_BORDERS by
+ duplicating last PS parameters and incrementing noEnv */
+ if (pBsData->aEnvStartStop[pBsData->noEnv] < h_ps_d->noSubSamples) {
+ for (gr = 0; gr < NO_HI_RES_IID_BINS; gr++) {
+ pBsData->aaIidIndex[pBsData->noEnv][gr] =
+ pBsData->aaIidIndex[pBsData->noEnv-1][gr];
+ }
+ for (gr = 0; gr < NO_HI_RES_ICC_BINS; gr++) {
+ pBsData->aaIccIndex[pBsData->noEnv][gr] =
+ pBsData->aaIccIndex[pBsData->noEnv-1][gr];
+ }
+ pBsData->noEnv++;
+ pBsData->aEnvStartStop[pBsData->noEnv] = h_ps_d->noSubSamples;
+ }
+
+ /* enforce strictly monotonic increasing borders */
+ for (env=1; env<pBsData->noEnv; env++) {
+ UCHAR thr;
+ thr = (UCHAR)h_ps_d->noSubSamples - (pBsData->noEnv - env);
+ if (pBsData->aEnvStartStop[env] > thr) {
+ pBsData->aEnvStartStop[env] = thr;
+ }
+ else {
+ thr = pBsData->aEnvStartStop[env-1]+1;
+ if (pBsData->aEnvStartStop[env] < thr) {
+ pBsData->aEnvStartStop[env] = thr;
+ }
+ }
+ }
+ } /* if (h_ps_d->bFrameClass == 0) ... else */
+
+ /* copy data prior to possible 20<->34 in-place mapping */
+ for (env=0; env<pBsData->noEnv; env++) {
+ UCHAR i;
+ for (i=0; i<NO_HI_RES_IID_BINS; i++) {
+ h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][i] = pBsData->aaIidIndex[env][i];
+ }
+ for (i=0; i<NO_HI_RES_ICC_BINS; i++) {
+ h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][i] = pBsData->aaIccIndex[env][i];
+ }
+ }
+
+
+ /* MPEG baseline PS */
+ /* Baseline version of PS always uses the hybrid filter structure with 20 stereo bands. */
+ /* If ICC/IID parameters for 34 stereo bands are decoded they have to be mapped to 20 */
+ /* stereo bands. */
+ /* Additionaly the IPD/OPD parameters won't be used. */
+
+ for (env=0; env<pBsData->noEnv; env++) {
+ if (pBsData->freqResIid == 2)
+ map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env], NO_HI_RES_IID_BINS);
+ if (pBsData->freqResIcc == 2)
+ map34IndexTo20 (h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env], NO_HI_RES_ICC_BINS);
+
+ /* IPD/OPD is disabled in baseline version and thus was removed here */
+ }
+
+ return (1);
+}
+
+
+/***************************************************************************/
+/*!
+
+ \brief Reads parametric stereo data from bitstream
+
+ \return
+
+****************************************************************************/
+unsigned int
+ReadPsData (HANDLE_PS_DEC h_ps_d, /*!< handle to struct PS_DEC */
+ HANDLE_FDK_BITSTREAM hBitBuf, /*!< handle to struct BIT_BUF */
+ int nBitsLeft /*!< max number of bits available */
+ )
+{
+ MPEG_PS_BS_DATA *pBsData;
+
+ UCHAR gr, env;
+ SCHAR dtFlag;
+ INT startbits;
+ Huffman CurrentTable;
+ SCHAR bEnableHeader;
+
+ if (!h_ps_d)
+ return 0;
+
+ pBsData = &h_ps_d->bsData[h_ps_d->bsReadSlot].mpeg;
+
+ if (h_ps_d->bsReadSlot != h_ps_d->bsLastSlot) {
+ /* Copy last header data */
+ FDKmemcpy(pBsData, &h_ps_d->bsData[h_ps_d->bsLastSlot].mpeg, sizeof(MPEG_PS_BS_DATA));
+ }
+
+
+ startbits = (INT) FDKgetValidBits(hBitBuf);
+
+ bEnableHeader = (SCHAR) FDKreadBits (hBitBuf, 1);
+
+ /* Read header */
+ if (bEnableHeader) {
+ pBsData->bPsHeaderValid = 1;
+ pBsData->bEnableIid = (UCHAR) FDKreadBits (hBitBuf, 1);
+ if (pBsData->bEnableIid) {
+ pBsData->modeIid = (UCHAR) FDKreadBits (hBitBuf, 3);
+ }
+
+ pBsData->bEnableIcc = (UCHAR) FDKreadBits (hBitBuf, 1);
+ if (pBsData->bEnableIcc) {
+ pBsData->modeIcc = (UCHAR) FDKreadBits (hBitBuf, 3);
+ }
+
+ pBsData->bEnableExt = (UCHAR) FDKreadBits (hBitBuf, 1);
+ }
+
+ pBsData->bFrameClass = (UCHAR) FDKreadBits (hBitBuf, 1);
+ if (pBsData->bFrameClass == 0) {
+ /* FIX_BORDERS NoEnv=0,1,2,4 */
+ pBsData->noEnv = FDK_sbrDecoder_aFixNoEnvDecode[(UCHAR) FDKreadBits (hBitBuf, 2)];
+ /* all additional handling of env borders is now in DecodePs() */
+ }
+ else {
+ /* VAR_BORDERS NoEnv=1,2,3,4 */
+ pBsData->noEnv = 1+(UCHAR) FDKreadBits (hBitBuf, 2);
+ for (env=1; env<pBsData->noEnv+1; env++)
+ pBsData->aEnvStartStop[env] = ((UCHAR) FDKreadBits (hBitBuf, 5)) + 1;
+ /* all additional handling of env borders is now in DecodePs() */
+ }
+
+ /* verify that IID & ICC modes (quant grid, freq res) are supported */
+ if ((pBsData->modeIid > 5) || (pBsData->modeIcc > 5)) {
+ /* no useful PS data could be read from bitstream */
+ h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_none;
+ /* discard all remaining bits */
+ nBitsLeft -= startbits - FDKgetValidBits(hBitBuf);
+ while (nBitsLeft) {
+ int i = nBitsLeft;
+ if (i>8) {
+ i = 8;
+ }
+ FDKreadBits (hBitBuf, i);
+ nBitsLeft -= i;
+ }
+ return (startbits - FDKgetValidBits(hBitBuf));
+ }
+
+ if (pBsData->modeIid > 2){
+ pBsData->freqResIid = pBsData->modeIid-3;
+ pBsData->bFineIidQ = 1;
+ }
+ else{
+ pBsData->freqResIid = pBsData->modeIid;
+ pBsData->bFineIidQ = 0;
+ }
+
+ if (pBsData->modeIcc > 2){
+ pBsData->freqResIcc = pBsData->modeIcc-3;
+ }
+ else{
+ pBsData->freqResIcc = pBsData->modeIcc;
+ }
+
+
+ /* Extract IID data */
+ if (pBsData->bEnableIid) {
+ for (env=0; env<pBsData->noEnv; env++) {
+ dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1);
+ if (!dtFlag)
+ {
+ if (pBsData->bFineIidQ)
+ CurrentTable = (Huffman)&aBookPsIidFineFreqDecode;
+ else
+ CurrentTable = (Huffman)&aBookPsIidFreqDecode;
+ }
+ else
+ {
+ if (pBsData->bFineIidQ)
+ CurrentTable = (Huffman)&aBookPsIidFineTimeDecode;
+ else
+ CurrentTable = (Huffman)&aBookPsIidTimeDecode;
+ }
+
+ for (gr = 0; gr < FDK_sbrDecoder_aNoIidBins[pBsData->freqResIid]; gr++)
+ pBsData->aaIidIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL);
+ pBsData->abIidDtFlag[env] = dtFlag;
+ }
+ }
+
+ /* Extract ICC data */
+ if (pBsData->bEnableIcc) {
+ for (env=0; env<pBsData->noEnv; env++) {
+ dtFlag = (SCHAR)FDKreadBits (hBitBuf, 1);
+ if (!dtFlag)
+ CurrentTable = (Huffman)&aBookPsIccFreqDecode;
+ else
+ CurrentTable = (Huffman)&aBookPsIccTimeDecode;
+
+ for (gr = 0; gr < FDK_sbrDecoder_aNoIccBins[pBsData->freqResIcc]; gr++)
+ pBsData->aaIccIndex[env][gr] = decode_huff_cw(CurrentTable,hBitBuf,NULL);
+ pBsData->abIccDtFlag[env] = dtFlag;
+ }
+ }
+
+ if (pBsData->bEnableExt) {
+
+ /*!
+ Decoders that support only the baseline version of the PS tool are allowed
+ to ignore the IPD/OPD data, but according header data has to be parsed.
+ ISO/IEC 14496-3 Subpart 8 Annex 4
+ */
+
+ int cnt = FDKreadBits(hBitBuf, PS_EXTENSION_SIZE_BITS);
+ if (cnt == (1<<PS_EXTENSION_SIZE_BITS)-1) {
+ cnt += FDKreadBits(hBitBuf, PS_EXTENSION_ESC_COUNT_BITS);
+ }
+ while (cnt--)
+ FDKreadBits(hBitBuf, 8);
+ }
+
+
+ /* new PS data was read from bitstream */
+ h_ps_d->bPsDataAvail[h_ps_d->bsReadSlot] = ppt_mpeg;
+
+
+
+ return (startbits - FDKgetValidBits(hBitBuf));
+}
+
diff --git a/libSBRdec/src/psbitdec.h b/libSBRdec/src/psbitdec.h
new file mode 100644
index 0000000..3b65468
--- /dev/null
+++ b/libSBRdec/src/psbitdec.h
@@ -0,0 +1,103 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+#ifndef __PSBITDEC_H
+#define __PSBITDEC_H
+
+#include "sbrdecoder.h"
+
+
+#include "psdec.h"
+
+
+unsigned int
+ReadPsData (struct PS_DEC *h_ps_d,
+ HANDLE_FDK_BITSTREAM hBs,
+ int nBitsLeft);
+
+int
+DecodePs(struct PS_DEC *h_ps_d,
+ const UCHAR frameError);
+
+
+#endif /* __PSBITDEC_H */
diff --git a/libSBRdec/src/psdec.cpp b/libSBRdec/src/psdec.cpp
new file mode 100644
index 0000000..d494c65
--- /dev/null
+++ b/libSBRdec/src/psdec.cpp
@@ -0,0 +1,1414 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief parametric stereo decoder
+*/
+
+#include "psdec.h"
+
+
+
+#include "FDK_bitbuffer.h"
+#include "psdec_hybrid.h"
+
+#include "sbr_rom.h"
+#include "sbr_ram.h"
+
+#include "FDK_tools_rom.h"
+
+#include "genericStds.h"
+
+#include "FDK_trigFcts.h"
+
+
+/********************************************************************/
+/* MLQUAL DEFINES */
+/********************************************************************/
+
+ #define FRACT_ZERO FRACT_BITS-1
+/********************************************************************/
+
+SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d );
+
+void ResetPsDeCor( HANDLE_PS_DEC h_ps_d );
+
+
+/***** HELPERS *****/
+
+static void assignTimeSlotsPS (FIXP_DBL *bufAdr, FIXP_DBL **bufPtr, const int numSlots, const int numChan);
+
+
+
+/*******************/
+
+#define DIV3 FL2FXCONST_DBL(1.f/3.f) /* division 3.0 */
+#define DIV1_5 FL2FXCONST_DBL(2.f/3.f) /* division 1.5 */
+
+/***************************************************************************/
+/*!
+ \brief Creates one instance of the PS_DEC struct
+
+ \return Error info
+
+****************************************************************************/
+int
+CreatePsDec( HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */
+ int aacSamplesPerFrame
+ )
+{
+ SBR_ERROR errorInfo = SBRDEC_OK;
+ HANDLE_PS_DEC h_ps_d;
+ int i;
+
+ if (*h_PS_DEC == NULL) {
+ /* Get ps dec ram */
+ h_ps_d = GetRam_ps_dec();
+ if (h_ps_d == NULL) {
+ errorInfo = SBRDEC_MEM_ALLOC_FAILED;
+ goto bail;
+ }
+ } else {
+ /* Reset an open instance */
+ h_ps_d = *h_PS_DEC;
+ }
+
+ /* initialisation */
+ switch (aacSamplesPerFrame) {
+ case 960:
+ h_ps_d->noSubSamples = 30; /* col */
+ break;
+ case 1024:
+ h_ps_d->noSubSamples = 32; /* col */
+ break;
+ default:
+ h_ps_d->noSubSamples = -1;
+ break;
+ }
+
+ if (h_ps_d->noSubSamples > MAX_NUM_COL
+ || h_ps_d->noSubSamples <= 0)
+ {
+ goto bail;
+ }
+ h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */
+
+ h_ps_d->psDecodedPrv = 0;
+ h_ps_d->procFrameBased = -1;
+ for (i = 0; i < (1)+1; i++) {
+ h_ps_d->bPsDataAvail[i] = ppt_none;
+ }
+
+
+ for (i = 0; i < (1)+1; i++) {
+ FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA));
+ }
+
+ errorInfo = ResetPsDec( h_ps_d );
+
+ if ( errorInfo != SBRDEC_OK )
+ goto bail;
+
+ ResetPsDeCor( h_ps_d );
+
+ *h_PS_DEC = h_ps_d;
+
+
+
+ return 0;
+
+bail:
+ DeletePsDec(&h_ps_d);
+
+ return -1;
+} /*END CreatePsDec */
+
+/***************************************************************************/
+/*!
+ \brief Delete one instance of the PS_DEC struct
+
+ \return Error info
+
+****************************************************************************/
+int
+DeletePsDec( HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */
+{
+ if (*h_PS_DEC == NULL) {
+ return -1;
+ }
+
+
+ FreeRam_ps_dec(h_PS_DEC);
+
+
+ return 0;
+} /*END DeletePsDec */
+
+/***************************************************************************/
+/*!
+ \brief resets some values of the PS handle to default states
+
+ \return
+
+****************************************************************************/
+SBR_ERROR ResetPsDec( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */
+{
+ SBR_ERROR errorInfo = SBRDEC_OK;
+ INT i;
+
+ const UCHAR noQmfBandsInHybrid20 = 3;
+ /* const UCHAR noQmfBandsInHybrid34 = 5; */
+
+ const UCHAR aHybridResolution20[] = { HYBRID_8_CPLX,
+ HYBRID_2_REAL,
+ HYBRID_2_REAL };
+
+ h_ps_d->specificTo.mpeg.delayBufIndex = 0;
+
+ /* explicitly init state variables to safe values (until first ps header arrives) */
+
+ h_ps_d->specificTo.mpeg.lastUsb = 0;
+
+ h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = -(DFRACT_BITS-1);
+
+ FDKmemclear(h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf, (NO_QMF_CHANNELS-FIRST_DELAY_SB)*sizeof(UCHAR));
+ h_ps_d->specificTo.mpeg.noSampleDelay = delayIndexQmf[0];
+
+ for (i=0 ; i < NO_SERIAL_ALLPASS_LINKS; i++) {
+ h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[i] = 0;
+ }
+
+ h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufReal;
+
+ assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB),
+ &h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[1],
+ h_ps_d->specificTo.mpeg.noSampleDelay-1,
+ (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB));
+
+ h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] = h_ps_d->specificTo.mpeg.aaQmfDelayBufImag;
+
+ assignTimeSlotsPS ( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0] + (NO_QMF_CHANNELS-FIRST_DELAY_SB),
+ &h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[1],
+ h_ps_d->specificTo.mpeg.noSampleDelay-1,
+ (NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB));
+
+ /* Hybrid Filter Bank 1 creation. */
+ errorInfo = InitHybridFilterBank ( &h_ps_d->specificTo.mpeg.hybrid,
+ h_ps_d->noSubSamples,
+ noQmfBandsInHybrid20,
+ aHybridResolution20 );
+
+ for ( i = 0; i < NO_IID_GROUPS; i++ )
+ {
+ h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f);
+ h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f);
+ }
+
+ FDKmemclear( h_ps_d->specificTo.mpeg.h21rPrev, sizeof( h_ps_d->specificTo.mpeg.h21rPrev ) );
+ FDKmemclear( h_ps_d->specificTo.mpeg.h22rPrev, sizeof( h_ps_d->specificTo.mpeg.h22rPrev ) );
+
+ return errorInfo;
+}
+
+/***************************************************************************/
+/*!
+ \brief clear some buffers used in decorrelation process
+
+ \return
+
+****************************************************************************/
+void ResetPsDeCor( HANDLE_PS_DEC h_ps_d ) /*!< pointer to the module state */
+{
+ INT i;
+
+ FDKmemclear(h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aPowerPrevScal, NO_MID_RES_BINS*sizeof(SCHAR));
+
+ for (i=0 ; i < FIRST_DELAY_SB ; i++) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ }
+ for (i=0 ; i < NO_SUB_QMF_CHANNELS ; i++) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ }
+
+}
+
+/*******************************************************************************/
+
+/* slot based funcion prototypes */
+
+static void deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d,
+
+ FIXP_DBL *mHybridRealLeft,
+ FIXP_DBL *mHybridImagLeft,
+ SCHAR sf_mHybridLeft,
+
+ FIXP_DBL *rIntBufferLeft,
+ FIXP_DBL *iIntBufferLeft,
+ SCHAR sf_IntBuffer,
+
+ FIXP_DBL *mHybridRealRight,
+ FIXP_DBL *mHybridImagRight,
+
+ FIXP_DBL *rIntBufferRight,
+ FIXP_DBL *iIntBufferRight );
+
+static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d,
+
+ FIXP_DBL *mHybridRealLeft,
+ FIXP_DBL *mHybridImagLeft,
+
+ FIXP_DBL *QmfLeftReal,
+ FIXP_DBL *QmfLeftImag,
+
+ FIXP_DBL *mHybridRealRight,
+ FIXP_DBL *mHybridImagRight,
+
+ FIXP_DBL *QmfRightReal,
+ FIXP_DBL *QmfRightImag
+ );
+
+
+/***************************************************************************/
+/*!
+ \brief Get scale factor for all ps delay buffer.
+
+ \return
+
+****************************************************************************/
+static
+int getScaleFactorPsStatesBuffer(HANDLE_PS_DEC h_ps_d)
+{
+ INT i;
+ int scale = DFRACT_BITS-1;
+
+ for (i=0; i<NO_QMF_BANDS_HYBRID20; i++) {
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS));
+ }
+
+ for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB));
+ }
+
+ for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS));
+ }
+
+ for (i=0; i<FIRST_DELAY_SB; i++) {
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS));
+ }
+
+ for (i=0; i<NO_SUB_QMF_CHANNELS; i++) {
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS));
+ }
+
+ for (i=0; i<MAX_DELAY_BUFFER_SIZE; i++)
+ {
+ INT len;
+ if (i==0)
+ len = NO_QMF_CHANNELS-FIRST_DELAY_SB;
+ else
+ len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB;
+
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[i], len));
+ scale = fMin(scale, getScalefactor(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len));
+ }
+
+ return (scale);
+}
+
+/***************************************************************************/
+/*!
+ \brief Rescale all ps delay buffer.
+
+ \return
+
+****************************************************************************/
+static
+void scalePsStatesBuffer(HANDLE_PS_DEC h_ps_d,
+ int scale)
+{
+ INT i;
+
+ if (scale < 0)
+ scale = fixMax((INT)scale,(INT)-(DFRACT_BITS-1));
+ else
+ scale = fixMin((INT)scale,(INT)DFRACT_BITS-1);
+
+ for (i=0; i<NO_QMF_BANDS_HYBRID20; i++) {
+ scaleValues( h_ps_d->specificTo.mpeg.hybrid.mQmfBufferRealSlot[i], NO_SUB_QMF_CHANNELS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.hybrid.mQmfBufferImagSlot[i], NO_SUB_QMF_CHANNELS, scale );
+ }
+
+ for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+ scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[i], FIRST_DELAY_SB, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[i], FIRST_DELAY_SB, scale );
+ }
+
+ for (i=0; i<NO_SAMPLE_DELAY_ALLPASS; i++) {
+ scaleValues( h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[i], NO_SUB_QMF_CHANNELS, scale );
+ }
+
+ for (i=0; i<FIRST_DELAY_SB; i++) {
+ scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+ }
+
+ for (i=0; i<NO_SUB_QMF_CHANNELS; i++) {
+ scaleValues( h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[i], NO_DELAY_LENGTH_VECTORS, scale );
+ }
+
+ for (i=0; i<MAX_DELAY_BUFFER_SIZE; i++) {
+ INT len;
+ if (i==0)
+ len = NO_QMF_CHANNELS-FIRST_DELAY_SB;
+ else
+ len = NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB;
+
+ scaleValues( h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[i], len, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[i], len, scale );
+ }
+
+ scale <<= 1;
+
+ scaleValues( h_ps_d->specificTo.mpeg.aPeakDecayFastBin, NO_MID_RES_BINS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aPrevPeakDiffBin, NO_MID_RES_BINS, scale );
+ scaleValues( h_ps_d->specificTo.mpeg.aPrevNrgBin, NO_MID_RES_BINS, scale );
+}
+
+/***************************************************************************/
+/*!
+ \brief Scale input channel to the same scalefactor and rescale hybrid
+ filterbank values
+
+ \return
+
+****************************************************************************/
+
+void scalFilterBankValues( HANDLE_PS_DEC h_ps_d,
+ FIXP_DBL **fixpQmfReal,
+ FIXP_DBL **fixpQmfImag,
+ int lsb,
+ int scaleFactorLowBandSplitLow,
+ int scaleFactorLowBandSplitHigh,
+ SCHAR *scaleFactorLowBand_lb,
+ SCHAR *scaleFactorLowBand_hb,
+ int scaleFactorHighBands,
+ INT *scaleFactorHighBand,
+ INT noCols
+ )
+{
+ INT maxScal;
+
+ INT i;
+
+ scaleFactorHighBands = -scaleFactorHighBands;
+ scaleFactorLowBandSplitLow = -scaleFactorLowBandSplitLow;
+ scaleFactorLowBandSplitHigh = -scaleFactorLowBandSplitHigh;
+
+ /* get max scale factor */
+ maxScal = fixMax(scaleFactorHighBands,fixMax(scaleFactorLowBandSplitLow, scaleFactorLowBandSplitHigh ));
+
+ {
+ int headroom = getScaleFactorPsStatesBuffer(h_ps_d);
+ maxScal = fixMax(maxScal,(INT)(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-headroom));
+ maxScal += 1;
+ }
+
+ /* scale whole left channel to the same scale factor */
+
+ /* low band ( overlap buffer ) */
+ if ( maxScal != scaleFactorLowBandSplitLow ) {
+ INT scale = scaleFactorLowBandSplitLow - maxScal;
+ for ( i=0; i<(6); i++ ) {
+ scaleValues( fixpQmfReal[i], lsb, scale );
+ scaleValues( fixpQmfImag[i], lsb, scale );
+ }
+ }
+ /* low band ( current frame ) */
+ if ( maxScal != scaleFactorLowBandSplitHigh ) {
+ INT scale = scaleFactorLowBandSplitHigh - maxScal;
+ /* for ( i=(6); i<(6)+MAX_NUM_COL; i++ ) { */
+ for ( i=(6); i<(6)+noCols; i++ ) {
+ scaleValues( fixpQmfReal[i], lsb, scale );
+ scaleValues( fixpQmfImag[i], lsb, scale );
+ }
+ }
+ /* high band */
+ if ( maxScal != scaleFactorHighBands ) {
+ INT scale = scaleFactorHighBands - maxScal;
+ /* for ( i=0; i<MAX_NUM_COL; i++ ) { */
+ for ( i=0; i<noCols; i++ ) {
+ scaleValues( &fixpQmfReal[i][lsb], (64)-lsb, scale );
+ scaleValues( &fixpQmfImag[i][lsb], (64)-lsb, scale );
+ }
+ }
+
+ if ( maxScal != h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer )
+ scalePsStatesBuffer(h_ps_d,(h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer-maxScal));
+
+ h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer = maxScal;
+ h_ps_d->specificTo.mpeg.scaleFactorPsDelayBuffer = maxScal;
+
+ *scaleFactorHighBand += maxScal - scaleFactorHighBands;
+
+ h_ps_d->rescal = maxScal - scaleFactorLowBandSplitHigh;
+ h_ps_d->sf_IntBuffer = maxScal;
+
+ *scaleFactorLowBand_lb += maxScal - scaleFactorLowBandSplitLow;
+ *scaleFactorLowBand_hb += maxScal - scaleFactorLowBandSplitHigh;
+}
+
+void rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
+ FIXP_DBL **QmfBufferReal, /* qmf filterbank values */
+ FIXP_DBL **QmfBufferImag, /* qmf filterbank values */
+ int lsb, /* sbr start subband */
+ INT noCols)
+{
+ int i;
+ /* scale back 6 timeslots look ahead for hybrid filterbank to original value */
+ for ( i=noCols; i<noCols + (6); i++ ) {
+ scaleValues( QmfBufferReal[i], lsb, h_ps_d->rescal );
+ scaleValues( QmfBufferImag[i], lsb, h_ps_d->rescal );
+ }
+}
+
+/***************************************************************************/
+/*!
+ \brief Generate decorrelated side channel using allpass/delay
+
+ \return
+
+****************************************************************************/
+static void
+deCorrelateSlotBased( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
+
+ FIXP_DBL *mHybridRealLeft, /*!< left (mono) hybrid values real */
+ FIXP_DBL *mHybridImagLeft, /*!< left (mono) hybrid values imag */
+ SCHAR sf_mHybridLeft, /*!< scalefactor for left (mono) hybrid bands */
+
+ FIXP_DBL *rIntBufferLeft, /*!< real qmf bands left (mono) (38x64) */
+ FIXP_DBL *iIntBufferLeft, /*!< real qmf bands left (mono) (38x64) */
+ SCHAR sf_IntBuffer, /*!< scalefactor for all left and right qmf bands */
+
+ FIXP_DBL *mHybridRealRight, /*!< right (decorrelated) hybrid values real */
+ FIXP_DBL *mHybridImagRight, /*!< right (decorrelated) hybrid values imag */
+
+ FIXP_DBL *rIntBufferRight, /*!< real qmf bands right (decorrelated) (38x64) */
+ FIXP_DBL *iIntBufferRight ) /*!< real qmf bands right (decorrelated) (38x64) */
+{
+
+ INT i, m, sb, gr, bin;
+
+ FIXP_DBL peakDiff, nrg, transRatio;
+
+ FIXP_DBL *RESTRICT aaLeftReal;
+ FIXP_DBL *RESTRICT aaLeftImag;
+
+ FIXP_DBL *RESTRICT aaRightReal;
+ FIXP_DBL *RESTRICT aaRightImag;
+
+ FIXP_DBL *RESTRICT pRealDelayBuffer;
+ FIXP_DBL *RESTRICT pImagDelayBuffer;
+
+ C_ALLOC_SCRATCH_START(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS);
+ C_ALLOC_SCRATCH_START(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS);
+
+/*!
+<pre>
+ parameter index qmf bands hybrid bands
+ ----------------------------------------------------------------------------
+ 0 0 0,7
+ 1 0 1,6
+ 2 0 2
+ 3 0 3 HYBRID BANDS
+ 4 1 9
+ 5 1 8
+ 6 2 10
+ 7 2 11
+ ----------------------------------------------------------------------------
+ 8 3
+ 9 4
+ 10 5
+ 11 6
+ 12 7
+ 13 8
+ 14 9,10 (2 ) QMF BANDS
+ 15 11 - 13 (3 )
+ 16 14 - 17 (4 )
+ 17 18 - 22 (5 )
+ 18 23 - 34 (12)
+ 19 35 - 63 (29)
+ ----------------------------------------------------------------------------
+</pre>
+*/
+
+ #define FLTR_SCALE 3
+
+ /* hybrid bands (parameter index 0 - 7) */
+ aaLeftReal = mHybridRealLeft;
+ aaLeftImag = mHybridImagLeft;
+
+ aaPowerSlot[0] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[0], aaLeftReal[0]), aaLeftImag[0], aaLeftImag[0] ) >> FLTR_SCALE ) +
+ ( fMultAddDiv2( fMultDiv2(aaLeftReal[7], aaLeftReal[7]), aaLeftImag[7], aaLeftImag[7] ) >> FLTR_SCALE );
+
+ aaPowerSlot[1] = ( fMultAddDiv2( fMultDiv2(aaLeftReal[1], aaLeftReal[1]), aaLeftImag[1], aaLeftImag[1] ) >> FLTR_SCALE ) +
+ ( fMultAddDiv2( fMultDiv2(aaLeftReal[6], aaLeftReal[6]), aaLeftImag[6], aaLeftImag[6] ) >> FLTR_SCALE );
+
+ aaPowerSlot[2] = fMultAddDiv2( fMultDiv2(aaLeftReal[2], aaLeftReal[2]), aaLeftImag[2], aaLeftImag[2] ) >> FLTR_SCALE;
+ aaPowerSlot[3] = fMultAddDiv2( fMultDiv2(aaLeftReal[3], aaLeftReal[3]), aaLeftImag[3], aaLeftImag[3] ) >> FLTR_SCALE;
+
+ aaPowerSlot[4] = fMultAddDiv2( fMultDiv2(aaLeftReal[9], aaLeftReal[9]), aaLeftImag[9], aaLeftImag[9] ) >> FLTR_SCALE;
+ aaPowerSlot[5] = fMultAddDiv2( fMultDiv2(aaLeftReal[8], aaLeftReal[8]), aaLeftImag[8], aaLeftImag[8] ) >> FLTR_SCALE;
+
+ aaPowerSlot[6] = fMultAddDiv2( fMultDiv2(aaLeftReal[10], aaLeftReal[10]), aaLeftImag[10], aaLeftImag[10] ) >> FLTR_SCALE;
+ aaPowerSlot[7] = fMultAddDiv2( fMultDiv2(aaLeftReal[11], aaLeftReal[11]), aaLeftImag[11], aaLeftImag[11] ) >> FLTR_SCALE;
+
+ /* qmf bands (parameter index 8 - 19) */
+ for ( bin = 8; bin < NO_MID_RES_BINS; bin++ ) {
+ FIXP_DBL slotNrg = FL2FXCONST_DBL(0.f);
+
+ for ( i = groupBorders20[bin+2]; i < groupBorders20[bin+3]; i++ ) { /* max loops: 29 */
+ slotNrg += fMultAddDiv2 ( fMultDiv2(rIntBufferLeft[i], rIntBufferLeft[i]), iIntBufferLeft[i], iIntBufferLeft[i]) >> FLTR_SCALE;
+ }
+ aaPowerSlot[bin] = slotNrg;
+
+ }
+
+
+ /* calculation of transient ratio */
+ for (bin=0; bin < NO_MID_RES_BINS; bin++) { /* noBins = 20 ( BASELINE_PS ) */
+
+ h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = fMult( h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin], PEAK_DECAY_FACTOR );
+
+ if (h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] < aaPowerSlot[bin]) {
+ h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] = aaPowerSlot[bin];
+ }
+
+ /* calculate PSmoothPeakDecayDiffNrg */
+ peakDiff = fMultAdd ( (h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]>>1),
+ INT_FILTER_COEFF, h_ps_d->specificTo.mpeg.aPeakDecayFastBin[bin] - aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin]);
+
+ /* save peakDiff for the next frame */
+ h_ps_d->specificTo.mpeg.aPrevPeakDiffBin[bin] = peakDiff;
+
+ nrg = h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] + fMult( INT_FILTER_COEFF, aaPowerSlot[bin] - h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] );
+
+ /* Negative energies don't exist. But sometimes they appear due to rounding. */
+
+ nrg = fixMax(nrg,FL2FXCONST_DBL(0.f));
+
+ /* save nrg for the next frame */
+ h_ps_d->specificTo.mpeg.aPrevNrgBin[bin] = nrg;
+
+ nrg = fMult( nrg, TRANSIENT_IMPACT_FACTOR );
+
+ /* save transient impact factor */
+ if ( peakDiff <= nrg || peakDiff == FL2FXCONST_DBL(0.0) ) {
+ aaTransRatioSlot[bin] = (FIXP_DBL)MAXVAL_DBL /* FL2FXCONST_DBL(1.0f)*/;
+ }
+ else if ( nrg <= FL2FXCONST_DBL(0.0f) ) {
+ aaTransRatioSlot[bin] = FL2FXCONST_DBL(0.f);
+ }
+ else {
+ /* scale to denominator */
+ INT scale_left = fixMax(0, CntLeadingZeros(peakDiff) - 1);
+ aaTransRatioSlot[bin] = schur_div( nrg<<scale_left, peakDiff<<scale_left, 16);
+ }
+ } /* bin */
+
+
+
+
+ #define DELAY_GROUP_OFFSET 20
+ #define NR_OF_DELAY_GROUPS 2
+
+ FIXP_DBL rTmp, iTmp, rTmp0, iTmp0, rR0, iR0;
+
+ INT TempDelay = h_ps_d->specificTo.mpeg.delayBufIndex; /* set delay indices */
+
+ pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferSubQmf[TempDelay];
+ pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferSubQmf[TempDelay];
+
+ aaLeftReal = mHybridRealLeft;
+ aaLeftImag = mHybridImagLeft;
+ aaRightReal = mHybridRealRight;
+ aaRightImag = mHybridImagRight;
+
+ /************************/
+ /* ICC groups : 0 - 9 */
+ /************************/
+
+ /* gr = ICC groups */
+ for (gr=0; gr < SUBQMF_GROUPS; gr++) {
+
+ transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+ /* sb = subQMF/QMF subband */
+ sb = groupBorders20[gr];
+
+ /* Update delay buffers, sample delay allpass = 2 */
+ rTmp0 = pRealDelayBuffer[sb];
+ iTmp0 = pImagDelayBuffer[sb];
+
+ pRealDelayBuffer[sb] = aaLeftReal[sb];
+ pImagDelayBuffer[sb] = aaLeftImag[sb];
+
+ /* delay by fraction */
+ cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReSubQmf20[sb], aaFractDelayPhaseFactorImSubQmf20[sb]);
+ rR0<<=1;
+ iR0<<=1;
+
+ FIXP_DBL *pAaaRealDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerSubQmf[sb];
+ FIXP_DBL *pAaaImagDelayRBufferSerSubQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerSubQmf[sb];
+
+ for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+
+ INT tmpDelayRSer = h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m];
+
+ /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */
+ rTmp0 = pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer];
+ iTmp0 = pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer];
+
+ /* delay by fraction */
+ cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReSubQmf20[sb][m], aaFractDelayPhaseFactorSerImSubQmf20[sb][m]);
+
+ rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], rR0)) << 1;
+ iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], iR0)) << 1;
+
+ pAaaRealDelayRBufferSerSubQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], rTmp);
+ pAaaImagDelayRBufferSerSubQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], iTmp);
+
+ rR0 = rTmp;
+ iR0 = iTmp;
+
+ pAaaRealDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m];
+ pAaaImagDelayRBufferSerSubQmf += aAllpassLinkDelaySer[m];
+
+ } /* m */
+
+ /* duck if a past transient is found */
+ aaRightReal[sb] = fMult(transRatio, rR0);
+ aaRightImag[sb] = fMult(transRatio, iR0);
+
+ } /* gr */
+
+
+ scaleValues( mHybridRealLeft, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+ scaleValues( mHybridImagLeft, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+ scaleValues( mHybridRealRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+ scaleValues( mHybridImagRight, NO_SUB_QMF_CHANNELS, -SCAL_HEADROOM );
+
+
+ /************************/
+
+ aaLeftReal = rIntBufferLeft;
+ aaLeftImag = iIntBufferLeft;
+ aaRightReal = rIntBufferRight;
+ aaRightImag = iIntBufferRight;
+
+ pRealDelayBuffer = h_ps_d->specificTo.mpeg.aaRealDelayBufferQmf[TempDelay];
+ pImagDelayBuffer = h_ps_d->specificTo.mpeg.aaImagDelayBufferQmf[TempDelay];
+
+ /************************/
+ /* ICC groups : 10 - 19 */
+ /************************/
+
+
+ /* gr = ICC groups */
+ for (gr=SUBQMF_GROUPS; gr < NO_IID_GROUPS - NR_OF_DELAY_GROUPS; gr++) {
+
+ transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+ /* sb = subQMF/QMF subband */
+ for (sb = groupBorders20[gr]; sb < groupBorders20[gr+1]; sb++) {
+ FIXP_DBL resR, resI;
+
+ /* decayScaleFactor = 1.0f + decay_cutoff * DECAY_SLOPE - DECAY_SLOPE * sb; DECAY_SLOPE = 0.05 */
+ FIXP_DBL decayScaleFactor = decayScaleFactTable[sb];
+
+ /* Update delay buffers, sample delay allpass = 2 */
+ rTmp0 = pRealDelayBuffer[sb];
+ iTmp0 = pImagDelayBuffer[sb];
+
+ pRealDelayBuffer[sb] = aaLeftReal[sb];
+ pImagDelayBuffer[sb] = aaLeftImag[sb];
+
+ /* delay by fraction */
+ cplxMultDiv2(&rR0, &iR0, rTmp0, iTmp0, aaFractDelayPhaseFactorReQmf[sb], aaFractDelayPhaseFactorImQmf[sb]);
+ rR0<<=1;
+ iR0<<=1;
+
+ resR = fMult(decayScaleFactor, rR0);
+ resI = fMult(decayScaleFactor, iR0);
+
+ FIXP_DBL *pAaaRealDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[sb];
+ FIXP_DBL *pAaaImagDelayRBufferSerQmf = h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[sb];
+
+ for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+
+ INT tmpDelayRSer = h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m];
+
+ /* get delayed values from according buffer : m(0)=3; m(1)=4; m(2)=5; */
+ rTmp0 = pAaaRealDelayRBufferSerQmf[tmpDelayRSer];
+ iTmp0 = pAaaImagDelayRBufferSerQmf[tmpDelayRSer];
+
+ /* delay by fraction */
+ cplxMultDiv2(&rTmp, &iTmp, rTmp0, iTmp0, aaFractDelayPhaseFactorSerReQmf[sb][m], aaFractDelayPhaseFactorSerImQmf[sb][m]);
+
+ rTmp = (rTmp - fMultDiv2(aAllpassLinkDecaySer[m], resR))<<1;
+ iTmp = (iTmp - fMultDiv2(aAllpassLinkDecaySer[m], resI))<<1;
+
+ resR = fMult(decayScaleFactor, rTmp);
+ resI = fMult(decayScaleFactor, iTmp);
+
+ pAaaRealDelayRBufferSerQmf[tmpDelayRSer] = rR0 + fMult(aAllpassLinkDecaySer[m], resR);
+ pAaaImagDelayRBufferSerQmf[tmpDelayRSer] = iR0 + fMult(aAllpassLinkDecaySer[m], resI);
+
+ rR0 = rTmp;
+ iR0 = iTmp;
+
+ pAaaRealDelayRBufferSerQmf += aAllpassLinkDelaySer[m];
+ pAaaImagDelayRBufferSerQmf += aAllpassLinkDelaySer[m];
+
+ } /* m */
+
+ /* duck if a past transient is found */
+ aaRightReal[sb] = fMult(transRatio, rR0);
+ aaRightImag[sb] = fMult(transRatio, iR0);
+
+ } /* sb */
+ } /* gr */
+
+ /************************/
+ /* ICC groups : 20, 21 */
+ /************************/
+
+
+ /* gr = ICC groups */
+ for (gr=DELAY_GROUP_OFFSET; gr < NO_IID_GROUPS; gr++) {
+
+ INT sbStart = groupBorders20[gr];
+ INT sbStop = groupBorders20[gr+1];
+
+ UCHAR *pDelayBufIdx = &h_ps_d->specificTo.mpeg.aDelayBufIndexDelayQmf[sbStart-FIRST_DELAY_SB];
+
+ transRatio = aaTransRatioSlot[bins2groupMap20[gr]];
+
+ /* sb = subQMF/QMF subband */
+ for (sb = sbStart; sb < sbStop; sb++) {
+
+ /* Update delay buffers */
+ rR0 = h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB];
+ iR0 = h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB];
+
+ h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftReal[sb];
+ h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[*pDelayBufIdx][sb-FIRST_DELAY_SB] = aaLeftImag[sb];
+
+ /* duck if a past transient is found */
+ aaRightReal[sb] = fMult(transRatio, rR0);
+ aaRightImag[sb] = fMult(transRatio, iR0);
+
+ if (++(*pDelayBufIdx) >= delayIndexQmf[sb]) {
+ *pDelayBufIdx = 0;
+ }
+ pDelayBufIdx++;
+
+ } /* sb */
+ } /* gr */
+
+
+ /* Update delay buffer index */
+ if (++h_ps_d->specificTo.mpeg.delayBufIndex >= NO_SAMPLE_DELAY_ALLPASS)
+ h_ps_d->specificTo.mpeg.delayBufIndex = 0;
+
+ for (m=0; m<NO_SERIAL_ALLPASS_LINKS ; m++) {
+ if (++h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m] >= aAllpassLinkDelaySer[m])
+ h_ps_d->specificTo.mpeg.aDelayRBufIndexSer[m] = 0;
+ }
+
+
+ scaleValues( &rIntBufferLeft[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+ scaleValues( &iIntBufferLeft[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+ scaleValues( &rIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+ scaleValues( &iIntBufferRight[NO_QMF_BANDS_HYBRID20], NO_QMF_CHANNELS-NO_QMF_BANDS_HYBRID20, -SCAL_HEADROOM );
+
+ /* free memory on scratch */
+ C_ALLOC_SCRATCH_END(aaTransRatioSlot, FIXP_DBL, NO_MID_RES_BINS);
+ C_ALLOC_SCRATCH_END(aaPowerSlot, FIXP_DBL, NO_MID_RES_BINS);
+}
+
+
+void initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
+ int env,
+ int usb
+ ) {
+
+ INT group = 0;
+ INT bin = 0;
+ INT noIidSteps;
+
+/* const UCHAR *pQuantizedIIDs;*/
+
+ FIXP_SGL invL;
+ FIXP_DBL ScaleL, ScaleR;
+ FIXP_DBL Alpha, Beta;
+ FIXP_DBL h11r, h12r, h21r, h22r;
+
+ const FIXP_DBL *PScaleFactors;
+
+ /* Overwrite old values in delay buffers when upper subband is higher than in last frame */
+ if (env == 0) {
+
+ if ((usb > h_ps_d->specificTo.mpeg.lastUsb) && h_ps_d->specificTo.mpeg.lastUsb) {
+
+ INT i,k,length;
+
+ for (i=h_ps_d->specificTo.mpeg.lastUsb ; i < FIRST_DELAY_SB; i++) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaRealDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ FDKmemclear(h_ps_d->specificTo.mpeg.aaaImagDelayRBufferSerQmf[i], NO_DELAY_LENGTH_VECTORS*sizeof(FIXP_DBL));
+ }
+
+ for (k=0 ; k<NO_SAMPLE_DELAY_ALLPASS; k++) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[k], FIRST_DELAY_SB*sizeof(FIXP_DBL));
+ }
+ length = (usb-FIRST_DELAY_SB)*sizeof(FIXP_DBL);
+ if(length>0) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[0], length);
+ FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[0], length);
+ }
+ length = (fixMin(NO_DELAY_BUFFER_BANDS,(INT)usb)-FIRST_DELAY_SB)*sizeof(FIXP_DBL);
+ if(length>0) {
+ for (k=1 ; k < h_ps_d->specificTo.mpeg.noSampleDelay; k++) {
+ FDKmemclear(h_ps_d->specificTo.mpeg.pAaRealDelayBufferQmf[k], length);
+ FDKmemclear(h_ps_d->specificTo.mpeg.pAaImagDelayBufferQmf[k], length);
+ }
+ }
+ }
+ h_ps_d->specificTo.mpeg.lastUsb = usb;
+ } /* env == 0 */
+
+ if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ)
+ {
+ PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */
+ noIidSteps = NO_IID_STEPS_FINE;
+ /*pQuantizedIIDs = quantizedIIDsFine;*/
+ }
+
+ else
+ {
+ PScaleFactors = ScaleFactors; /* values are shiftet right by one */
+ noIidSteps = NO_IID_STEPS;
+ /*pQuantizedIIDs = quantizedIIDs;*/
+ }
+
+
+ /* dequantize and decode */
+ for ( group = 0; group < NO_IID_GROUPS; group++ ) {
+
+ bin = bins2groupMap20[group];
+
+ /*!
+ <h3> type 'A' rotation </h3>
+ mixing procedure R_a, used in baseline version<br>
+
+ Scale-factor vectors c1 and c2 are precalculated in initPsTables () and stored in
+ scaleFactors[] and scaleFactorsFine[] = pScaleFactors [].
+ From the linearized IID parameters (intensity differences), two scale factors are
+ calculated. They are used to obtain the coefficients h11... h22.
+ */
+
+ /* ScaleR and ScaleL are scaled by 1 shift right */
+
+ ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]];
+ ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.coef.aaIidIndexMapped[env][bin]];
+
+ Beta = fMult (fMult( Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]], ( ScaleR - ScaleL )), FIXP_SQRT05);
+ Alpha = Alphas[h_ps_d->specificTo.mpeg.coef.aaIccIndexMapped[env][bin]]>>1;
+
+ /* Alpha and Beta are now both scaled by 2 shifts right */
+
+ /* calculate the coefficients h11... h22 from scale-factors and ICC parameters */
+
+ /* h values are scaled by 1 shift right */
+ {
+ FIXP_DBL trigData[4];
+
+ inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData);
+ h11r = fMult( ScaleL, trigData[0]);
+ h12r = fMult( ScaleR, trigData[2]);
+ h21r = fMult( ScaleL, trigData[1]);
+ h22r = fMult( ScaleR, trigData[3]);
+ }
+ /*****************************************************************************************/
+ /* Interpolation of the matrices H11... H22: */
+ /* */
+ /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / (n[e+1] - n[e]) */
+ /* ... */
+ /*****************************************************************************************/
+
+ /* invL = 1/(length of envelope) */
+ invL = FX_DBL2FX_SGL(GetInvInt(h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env]));
+
+ h_ps_d->specificTo.mpeg.coef.H11r[group] = h_ps_d->specificTo.mpeg.h11rPrev[group];
+ h_ps_d->specificTo.mpeg.coef.H12r[group] = h_ps_d->specificTo.mpeg.h12rPrev[group];
+ h_ps_d->specificTo.mpeg.coef.H21r[group] = h_ps_d->specificTo.mpeg.h21rPrev[group];
+ h_ps_d->specificTo.mpeg.coef.H22r[group] = h_ps_d->specificTo.mpeg.h22rPrev[group];
+
+ h_ps_d->specificTo.mpeg.coef.DeltaH11r[group] = fMult ( h11r - h_ps_d->specificTo.mpeg.coef.H11r[group], invL );
+ h_ps_d->specificTo.mpeg.coef.DeltaH12r[group] = fMult ( h12r - h_ps_d->specificTo.mpeg.coef.H12r[group], invL );
+ h_ps_d->specificTo.mpeg.coef.DeltaH21r[group] = fMult ( h21r - h_ps_d->specificTo.mpeg.coef.H21r[group], invL );
+ h_ps_d->specificTo.mpeg.coef.DeltaH22r[group] = fMult ( h22r - h_ps_d->specificTo.mpeg.coef.H22r[group], invL );
+
+ /* update prev coefficients for interpolation in next envelope */
+
+ h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r;
+ h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r;
+ h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r;
+ h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r;
+
+ } /* group loop */
+}
+
+
+static void applySlotBasedRotation( HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */
+
+ FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */
+ FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */
+
+ FIXP_DBL *QmfLeftReal, /*!< real bands left qmf channel */
+ FIXP_DBL *QmfLeftImag, /*!< imag bands left qmf channel */
+
+ FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */
+ FIXP_DBL *mHybridImagRight, /*!< hybrid values imag right */
+
+ FIXP_DBL *QmfRightReal, /*!< real bands right qmf channel */
+ FIXP_DBL *QmfRightImag /*!< imag bands right qmf channel */
+ )
+{
+ INT group;
+ INT subband;
+
+ FIXP_DBL *RESTRICT HybrLeftReal;
+ FIXP_DBL *RESTRICT HybrLeftImag;
+ FIXP_DBL *RESTRICT HybrRightReal;
+ FIXP_DBL *RESTRICT HybrRightImag;
+
+ FIXP_DBL tmpLeft, tmpRight;
+
+
+ /**********************************************************************************************/
+ /*!
+ <h2> Mapping </h2>
+
+ The number of stereo bands that is actually used depends on the number of availble
+ parameters for IID and ICC:
+ <pre>
+ nr. of IID para.| nr. of ICC para. | nr. of Stereo bands
+ ----------------|------------------|-------------------
+ 10,20 | 10,20 | 20
+ 10,20 | 34 | 34
+ 34 | 10,20 | 34
+ 34 | 34 | 34
+ </pre>
+ In the case the number of parameters for IIS and ICC differs from the number of stereo
+ bands, a mapping from the lower number to the higher number of parameters is applied.
+ Index mapping of IID and ICC parameters is already done in psbitdec.cpp. Further mapping is
+ not needed here in baseline version.
+ **********************************************************************************************/
+
+ /************************************************************************************************/
+ /*!
+ <h2> Mixing </h2>
+
+ To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, n_[e+1] the
+ parameters at position n[e] and n[e+1] are required as well as the subband domain signals
+ s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] represents the start position for
+ envelope e. The border positions n[e] are handled in DecodePS().
+
+ The stereo sub subband signals are constructed as:
+ <pre>
+ l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
+ r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
+ </pre>
+ In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... h22(b) need to
+ be calculated first (b: parameter index). Depending on ICC mode either mixing procedure R_a
+ or R_b is used for that. For both procedures, the parameters for parameter position n[e+1]
+ is used.
+ ************************************************************************************************/
+
+
+ /************************************************************************************************/
+ /*!
+ <h2>Phase parameters </h2>
+ With disabled phase parameters (which is the case in baseline version), the H-matrices are
+ just calculated by:
+
+ <pre>
+ H11(k,n[e+1] = h11(b(k))
+ (...)
+ b(k): parameter index according to mapping table
+ </pre>
+
+ <h2>Processing of the samples in the sub subbands </h2>
+ this loop includes the interpolation of the coefficients Hxx
+ ************************************************************************************************/
+
+
+ /* loop thru all groups ... */
+ HybrLeftReal = mHybridRealLeft;
+ HybrLeftImag = mHybridImagLeft;
+ HybrRightReal = mHybridRealRight;
+ HybrRightImag = mHybridImagRight;
+
+ /******************************************************/
+ /* construct stereo sub subband signals according to: */
+ /* */
+ /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */
+ /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */
+ /******************************************************/
+ for ( group = 0; group < SUBQMF_GROUPS; group++ ) {
+
+ h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group];
+ h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group];
+ h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group];
+ h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group];
+
+ subband = groupBorders20[group];
+
+ tmpLeft = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]);
+ tmpRight = fMultAddDiv2( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]);
+ HybrLeftReal [subband] = tmpLeft<<1;
+ HybrRightReal[subband] = tmpRight<<1;
+
+ tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]);
+ tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]);
+ HybrLeftImag [subband] = tmpLeft;
+ HybrRightImag[subband] = tmpRight;
+ }
+
+ /* continue in the qmf buffers */
+ HybrLeftReal = QmfLeftReal;
+ HybrLeftImag = QmfLeftImag;
+ HybrRightReal = QmfRightReal;
+ HybrRightImag = QmfRightImag;
+
+ for (; group < NO_IID_GROUPS; group++ ) {
+
+ h_ps_d->specificTo.mpeg.coef.H11r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH11r[group];
+ h_ps_d->specificTo.mpeg.coef.H12r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH12r[group];
+ h_ps_d->specificTo.mpeg.coef.H21r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH21r[group];
+ h_ps_d->specificTo.mpeg.coef.H22r[group] += h_ps_d->specificTo.mpeg.coef.DeltaH22r[group];
+
+ for ( subband = groupBorders20[group]; subband < groupBorders20[group + 1]; subband++ )
+ {
+ tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightReal[subband]);
+ tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftReal[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightReal[subband]);
+ HybrLeftReal [subband] = tmpLeft;
+ HybrRightReal[subband] = tmpRight;
+
+ tmpLeft = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H11r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H21r[group], HybrRightImag[subband]);
+ tmpRight = fMultAdd( fMultDiv2(h_ps_d->specificTo.mpeg.coef.H12r[group], HybrLeftImag[subband]), h_ps_d->specificTo.mpeg.coef.H22r[group], HybrRightImag[subband]);
+ HybrLeftImag [subband] = tmpLeft;
+ HybrRightImag[subband] = tmpRight;
+
+ } /* subband */
+ }
+}
+
+
+/***************************************************************************/
+/*!
+ \brief Applies IID, ICC, IPD and OPD parameters to the current frame.
+
+ \return none
+
+****************************************************************************/
+void
+ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/
+ FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */
+ FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */
+ FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */
+ FIXP_DBL *iIntBufferRight /*!< imag bands right qmf channel (38x64) */
+ )
+{
+
+ /*!
+ The 64-band QMF representation of the monaural signal generated by the SBR tool
+ is used as input of the PS tool. After the PS processing, the outputs of the left
+ and right hybrid synthesis filterbanks are used to generate the stereo output
+ signal.
+
+ <pre>
+
+ ------------- ---------- -------------
+ | Hybrid | M_n[k,m] | | L_n[k,m] | Hybrid | l[n]
+ m[n] --->| analysis |--------->| |--------->| synthesis |----->
+ | filter bank | | | | filter bank |
+ ------------- | Stereo | -------------
+ | | recon- |
+ | | stuction |
+ \|/ | |
+ ------------- | |
+ | De- | D_n[k,m] | |
+ | correlation |--------->| |
+ ------------- | | -------------
+ | | R_n[k,m] | Hybrid | r[n]
+ | |--------->| synthesis |----->
+ IID, ICC ------------------------>| | | filter bank |
+ (IPD, OPD) ---------- -------------
+
+ m[n]: QMF represantation of the mono input
+ M_n[k,m]: (sub-)sub-band domain signals of the mono input
+ D_n[k,m]: decorrelated (sub-)sub-band domain signals
+ L_n[k,m]: (sub-)sub-band domain signals of the left output
+ R_n[k,m]: (sub-)sub-band domain signals of the right output
+ l[n],r[n]: left/right output signals
+
+ </pre>
+ */
+
+ /* get temporary hybrid qmf values of one timeslot */
+ C_ALLOC_SCRATCH_START(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_START(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_START(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_START(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+
+ SCHAR sf_IntBuffer = h_ps_d->sf_IntBuffer;
+
+ /* clear workbuffer */
+ FDKmemclear(hybridRealLeft, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+ FDKmemclear(hybridImagLeft, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+ FDKmemclear(hybridRealRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+ FDKmemclear(hybridImagRight, NO_SUB_QMF_CHANNELS*sizeof(FIXP_DBL));
+
+
+ /*!
+ Hybrid analysis filterbank:
+ The lower 3 (5) of the 64 QMF subbands are further split to provide better frequency resolution.
+ for PS processing.
+ For the 10 and 20 stereo bands configuration, the QMF band H_0(w) is split
+ up into 8 (sub-) sub-bands and the QMF bands H_1(w) and H_2(w) are spit into 2 (sub-)
+ 4th. (See figures 8.20 and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) )
+ */
+
+
+ if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing */
+ { /* fill hybrid delay buffer. */
+ h_ps_d->procFrameBased = 0;
+
+ fillHybridDelayLine( rIntBufferLeft,
+ iIntBufferLeft,
+ hybridRealLeft,
+ hybridImagLeft,
+ hybridRealRight,
+ hybridImagRight,
+ &h_ps_d->specificTo.mpeg.hybrid );
+ }
+
+ slotBasedHybridAnalysis ( rIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values */
+ iIntBufferLeft[HYBRID_FILTER_DELAY], /* qmf filterbank values */
+ hybridRealLeft, /* hybrid filterbank values */
+ hybridImagLeft, /* hybrid filterbank values */
+ &h_ps_d->specificTo.mpeg.hybrid); /* hybrid filterbank handle */
+
+
+ SCHAR hybridScal = h_ps_d->specificTo.mpeg.hybrid.sf_mQmfBuffer;
+
+
+ /*!
+ Decorrelation:
+ By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) are
+ converted into de-correlated (sub-)sub-band samples d_k(n).
+ - k: frequency in hybrid spectrum
+ - n: time index
+ */
+
+ deCorrelateSlotBased( h_ps_d, /* parametric stereo decoder handle */
+ hybridRealLeft, /* left hybrid time slot */
+ hybridImagLeft,
+ hybridScal, /* scale factor of left hybrid time slot */
+ rIntBufferLeft[0], /* left qmf time slot */
+ iIntBufferLeft[0],
+ sf_IntBuffer, /* scale factor of left and right qmf time slot */
+ hybridRealRight, /* right hybrid time slot */
+ hybridImagRight,
+ rIntBufferRight, /* right qmf time slot */
+ iIntBufferRight );
+
+
+
+ /*!
+ Stereo Processing:
+ The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according to
+ the stereo cues which are defined per stereo band.
+ */
+
+
+ applySlotBasedRotation( h_ps_d, /* parametric stereo decoder handle */
+ hybridRealLeft, /* left hybrid time slot */
+ hybridImagLeft,
+ rIntBufferLeft[0], /* left qmf time slot */
+ iIntBufferLeft[0],
+ hybridRealRight, /* right hybrid time slot */
+ hybridImagRight,
+ rIntBufferRight, /* right qmf time slot */
+ iIntBufferRight );
+
+
+
+
+ /*!
+ Hybrid synthesis filterbank:
+ The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the hybrid synthesis
+ filterbanks which are identical to the 64 complex synthesis filterbank of the SBR tool. The
+ input to the filterbank are slots of 64 QMF samples. For each slot the filterbank outputs one
+ block of 64 samples of one reconstructed stereo channel. The hybrid synthesis filterbank is
+ computed seperatly for the left and right channel.
+ */
+
+
+ /* left channel */
+ slotBasedHybridSynthesis ( hybridRealLeft, /* one timeslot of hybrid filterbank values */
+ hybridImagLeft,
+ rIntBufferLeft[0], /* one timeslot of qmf filterbank values */
+ iIntBufferLeft[0],
+ &h_ps_d->specificTo.mpeg.hybrid ); /* hybrid filterbank handle */
+
+ /* right channel */
+ slotBasedHybridSynthesis ( hybridRealRight, /* one timeslot of hybrid filterbank values */
+ hybridImagRight,
+ rIntBufferRight, /* one timeslot of qmf filterbank values */
+ iIntBufferRight,
+ &h_ps_d->specificTo.mpeg.hybrid ); /* hybrid filterbank handle */
+
+
+
+
+
+
+
+ /* free temporary hybrid qmf values of one timeslot */
+ C_ALLOC_SCRATCH_END(hybridImagRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_END(hybridRealRight, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_END(hybridImagLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+ C_ALLOC_SCRATCH_END(hybridRealLeft, FIXP_DBL, NO_SUB_QMF_CHANNELS);
+
+}/* END ApplyPsSlot */
+
+
+/***************************************************************************/
+/*!
+
+ \brief assigns timeslots to an array
+
+ \return
+
+****************************************************************************/
+
+static void assignTimeSlotsPS (FIXP_DBL *bufAdr,
+ FIXP_DBL **bufPtr,
+ const int numSlots,
+ const int numChan)
+{
+ FIXP_DBL *ptr;
+ int slot;
+ ptr = bufAdr;
+ for(slot=0; slot < numSlots; slot++) {
+ bufPtr [slot] = ptr;
+ ptr += numChan;
+ }
+}
+
diff --git a/libSBRdec/src/psdec.h b/libSBRdec/src/psdec.h
new file mode 100644
index 0000000..e3a0424
--- /dev/null
+++ b/libSBRdec/src/psdec.h
@@ -0,0 +1,352 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Sbr decoder
+*/
+#ifndef __PSDEC_H
+#define __PSDEC_H
+
+#include "sbrdecoder.h"
+
+
+
+/* This PS decoder implements the baseline version. So it always uses the */
+/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */
+/* synthesis. The baseline version has to support the complete PS bitstream */
+/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */
+/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */
+/* 20 stereo bands. */
+
+
+#include "FDK_bitstream.h"
+
+#include "psdec_hybrid.h"
+
+#define SCAL_HEADROOM ( 2 )
+
+#define PS_EXTENSION_SIZE_BITS ( 4 )
+#define PS_EXTENSION_ESC_COUNT_BITS ( 8 )
+
+#define NO_QMF_CHANNELS ( 64 )
+#define MAX_NUM_COL ( 32 )
+
+
+ #define NO_QMF_BANDS_HYBRID20 ( 3 )
+ #define NO_SUB_QMF_CHANNELS ( 12 )
+
+ #define NRG_INT_COEFF ( 0.75f )
+ #define INT_FILTER_COEFF (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF ))
+ #define PEAK_DECAY_FACTOR (FL2FXCONST_DBL( 0.765928338364649f ))
+ #define TRANSIENT_IMPACT_FACTOR (FL2FXCONST_DBL( 2.0 / 3.0 ))
+
+ #define NO_SERIAL_ALLPASS_LINKS ( 3 )
+ #define MAX_NO_PS_ENV ( 4 + 1 ) /* +1 needed for VAR_BORDER */
+
+ #define MAX_DELAY_BUFFER_SIZE ( 14 )
+ #define NO_DELAY_BUFFER_BANDS ( 35 )
+
+ #define NO_HI_RES_BINS ( 34 )
+ #define NO_MID_RES_BINS ( 20 )
+ #define NO_LOW_RES_BINS ( 10 )
+
+ #define FIRST_DELAY_SB ( 23 )
+ #define NO_SAMPLE_DELAY_ALLPASS ( 2 )
+ #define NO_DELAY_LENGTH_VECTORS ( 12 ) /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */
+
+ #define NO_HI_RES_IID_BINS ( NO_HI_RES_BINS )
+ #define NO_HI_RES_ICC_BINS ( NO_HI_RES_BINS )
+
+ #define NO_MID_RES_IID_BINS ( NO_MID_RES_BINS )
+ #define NO_MID_RES_ICC_BINS ( NO_MID_RES_BINS )
+
+ #define NO_LOW_RES_IID_BINS ( NO_LOW_RES_BINS )
+ #define NO_LOW_RES_ICC_BINS ( NO_LOW_RES_BINS )
+
+ #define SUBQMF_GROUPS ( 10 )
+ #define QMF_GROUPS ( 12 )
+
+ #define SUBQMF_GROUPS_HI_RES ( 32 )
+ #define QMF_GROUPS_HI_RES ( 18 )
+
+ #define NO_IID_GROUPS ( SUBQMF_GROUPS + QMF_GROUPS )
+ #define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + QMF_GROUPS_HI_RES )
+
+ #define NO_IID_STEPS ( 7 ) /* 1 .. + 7 */
+ #define NO_IID_STEPS_FINE ( 15 ) /* 1 .. +15 */
+ #define NO_ICC_STEPS ( 8 ) /* 0 .. + 7 */
+
+ #define NO_IID_LEVELS ( 2 * NO_IID_STEPS + 1 ) /* - 7 .. + 7 */
+ #define NO_IID_LEVELS_FINE ( 2 * NO_IID_STEPS_FINE + 1 ) /* -15 .. +15 */
+ #define NO_ICC_LEVELS ( NO_ICC_STEPS ) /* 0 .. + 7 */
+
+ #define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */
+
+ struct PS_DEC_COEFFICIENTS {
+
+ FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+
+ FIXP_DBL DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+ FIXP_DBL DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */
+
+ SCHAR aaIidIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all envelopes and all IID bins */
+ SCHAR aaIccIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all envelopes and all ICC bins */
+
+ };
+
+
+
+
+typedef enum {
+ ppt_none = 0,
+ ppt_mpeg = 1,
+ ppt_drm = 2
+} PS_PAYLOAD_TYPE;
+
+
+typedef struct {
+ UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */
+
+ UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */
+ UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */
+ UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext bit.
+ If it is set to %1 the IPD and OPD parameters are sent.
+ If it is disabled, i.e. %0, the extension layer is skipped. */
+
+ UCHAR modeIid; /*!< The configuration of IID parameters (number of bands and
+ quantisation grid, iid_quant) is determined by iid_mode. */
+ UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters
+ (number of bands and quantisation grid) is determined by
+ icc_mode. */
+
+ UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */
+ UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */
+
+ UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */
+
+ UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter
+ positions of the current frame are uniformly spaced
+ accross the frame or they are defined using the positions
+ described by border_position. */
+
+ UCHAR noEnv; /*!< The number of envelopes per frame */
+ UCHAR aEnvStartStop[MAX_NO_PS_ENV+1]; /*!< In case of variable parameter spacing the parameter
+ positions are determined by border_position */
+
+ SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 => freq */
+ SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 => freq */
+
+ SCHAR aaIidIndex[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and all IID bins */
+ SCHAR aaIccIndex[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and all ICC bins */
+
+} MPEG_PS_BS_DATA;
+
+
+
+struct PS_DEC {
+
+ SCHAR noSubSamples;
+ SCHAR noChannels;
+
+ SCHAR procFrameBased; /*!< Helper to detected switching from frame based to slot based
+ processing */
+
+ PS_PAYLOAD_TYPE bPsDataAvail[(1)+1]; /*!< set if new data available from bitstream */
+ UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */
+
+ /* helpers for frame delay line */
+ UCHAR bsLastSlot; /*!< Index of last read slot. */
+ UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */
+ UCHAR processSlot; /*!< Index of current slot for processing (need for add. delay). */
+
+
+ INT rescal;
+ INT sf_IntBuffer;
+
+ union { /* Bitstream data */
+ MPEG_PS_BS_DATA mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. */
+ } bsData[(1)+1];
+
+ shouldBeUnion { /* Static data */
+ struct {
+ SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for previous frame */
+ SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for previous frame */
+
+ UCHAR delayBufIndex; /*!< Pointer to where the latest sample is in buffer */
+ UCHAR noSampleDelay; /*!< How many QMF samples delay is used. */
+ UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */
+
+ UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS]; /*!< Delay buffer for reverb filter */
+ UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB]; /*!< Delay buffer for ICC group 20 & 21 */
+
+ SCHAR scaleFactorPsDelayBuffer; /*!< Scale factor for ps delay buffer */
+
+ /* hybrid filter bank delay lines */
+ FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
+ FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)];
+
+ FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Real part delay buffer */
+ FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Imaginary part delay buffer */
+
+ FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer */
+ FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/
+
+ FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Real part delay buffer */
+ FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Imaginary part delay buffer */
+
+ FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
+ FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
+
+ FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */
+ FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */
+
+ HYBRID hybrid; /*!< hybrid filter bank struct 1 or 2. */
+
+ FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS]; /*!< energy of previous frame */
+ FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS]; /*!< peak difference of previous frame */
+ FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS]; /*!< Saved max. peak decay value per bin */
+ SCHAR aPowerPrevScal[NO_MID_RES_BINS]; /*!< Last power value (each bin) of previous frame */
+
+ FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
+ FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
+ FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
+ FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */
+
+ PS_DEC_COEFFICIENTS coef; /*!< temporal coefficients (reusable scratch memory) */
+
+ } mpeg;
+
+ } specificTo;
+
+
+};
+
+typedef struct PS_DEC *HANDLE_PS_DEC;
+
+
+int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame);
+
+int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC);
+
+void
+scalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
+ FIXP_DBL **fixpQmfReal, /* qmf filterbank values */
+ FIXP_DBL **fixpQmfImag, /* qmf filterbank values */
+ int lsb, /* sbr start subband */
+ int scaleFactorLowBandSplitLow,
+ int scaleFactorLowBandSplitHigh,
+ SCHAR *scaleFactorLowBand_lb,
+ SCHAR *scaleFactorLowBand_hb,
+ int scaleFactorHighBands,
+ INT *scaleFactorHighBand,
+ INT noCols);
+
+void
+rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
+ FIXP_DBL **QmfBufferReal, /* qmf filterbank values */
+ FIXP_DBL **QmfBufferImag, /* qmf filterbank values */
+ int lsb, /* sbr start subband */
+ INT noCols);
+
+
+void
+initSlotBasedRotation( HANDLE_PS_DEC h_ps_d,
+ int env,
+ int usb);
+
+void
+ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */
+ FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */
+ FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */
+ FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */
+ FIXP_DBL *iIntBufferRight); /* imag values of right qmf timeslot */
+
+
+
+#endif /* __PSDEC_H */
diff --git a/libSBRdec/src/psdec_hybrid.cpp b/libSBRdec/src/psdec_hybrid.cpp
new file mode 100644
index 0000000..7fc2c0a
--- /dev/null
+++ b/libSBRdec/src/psdec_hybrid.cpp
@@ -0,0 +1,652 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+#include "psdec_hybrid.h"
+
+
+#include "fft.h"
+#include "sbr_ram.h"
+
+#include "FDK_tools_rom.h"
+#include "sbr_rom.h"
+
+/*******************************************************************************
+ Functionname: InitHybridFilterBank
+ *******************************************************************************
+
+ Description: Init one instance of HANDLE_HYBRID stuct
+
+ Arguments:
+
+ Return: none
+
+*******************************************************************************/
+
+
+SBR_ERROR
+InitHybridFilterBank ( HANDLE_HYBRID hs, /*!< Handle to HYBRID struct. */
+ SCHAR frameSize, /*!< Framesize (in Qmf súbband samples). */
+ SCHAR noBands, /*!< Number of Qmf bands for hybrid filtering. */
+ const UCHAR *pResolution ) /*!< Resolution in Qmf bands (length noBands). */
+{
+ SCHAR i;
+ UCHAR maxNoChannels = 0;
+
+ for (i = 0; i < noBands; i++) {
+ hs->pResolution[i] = pResolution[i];
+ if(pResolution[i] > maxNoChannels)
+ maxNoChannels = pResolution[i];
+ }
+
+ hs->nQmfBands = noBands;
+ hs->frameSize = frameSize;
+ hs->qmfBufferMove = HYBRID_FILTER_LENGTH - 1;
+
+ hs->sf_mQmfBuffer = 0;
+
+ return SBRDEC_OK;
+}
+
+/*******************************************************************************
+ Functionname: dualChannelFiltering
+ *******************************************************************************
+
+ Description: fast 2-channel real-valued filtering with 6-tap delay.
+
+ Arguments:
+
+ Return: none
+
+*******************************************************************************/
+
+/*!
+2 channel filter
+<pre>
+ Filter Coefs:
+ 0.0,
+ 0.01899487526049,
+ 0.0,
+ -0.07293139167538,
+ 0.0,
+ 0.30596630545168,
+ 0.5,
+ 0.30596630545168,
+ 0.0,
+ -0.07293139167538,
+ 0.0,
+ 0.01899487526049,
+ 0.0
+
+
+ Filter design:
+ h[q,n] = g[n] * cos(2pi/2 * q * (n-6) ); n = 0..12, q = 0,1;
+
+ -> h[0,n] = g[n] * 1;
+ -> h[1,n] = g[n] * pow(-1,n);
+</pre>
+*/
+
+static void slotBasedDualChannelFiltering( const FIXP_DBL *pQmfReal,
+ const FIXP_DBL *pQmfImag,
+
+ FIXP_DBL *mHybridReal,
+ FIXP_DBL *mHybridImag)
+{
+
+ FIXP_DBL t1, t3, t5, t6;
+
+ /* symmetric filter coefficients */
+
+ /* you don't have to shift the result after fMult because of p2_13_20 <= 0.5 */
+ t1 = fMultDiv2(p2_13_20[1] , ( (pQmfReal[1] >> 1) + (pQmfReal[11] >> 1)));
+ t3 = fMultDiv2(p2_13_20[3] , ( (pQmfReal[3] >> 1) + (pQmfReal[ 9] >> 1)));
+ t5 = fMultDiv2(p2_13_20[5] , ( (pQmfReal[5] >> 1) + (pQmfReal[ 7] >> 1)));
+ t6 = fMultDiv2(p2_13_20[6] , (pQmfReal[6] >> 1) );
+
+ mHybridReal[0] = (t1 + t3 + t5 + t6) << 2;
+ mHybridReal[1] = (- t1 - t3 - t5 + t6) << 2;
+
+ t1 = fMultDiv2(p2_13_20[1] , ( (pQmfImag[1] >> 1) + (pQmfImag[11] >> 1)));
+ t3 = fMultDiv2(p2_13_20[3] , ( (pQmfImag[3] >> 1) + (pQmfImag[ 9] >> 1)));
+ t5 = fMultDiv2(p2_13_20[5] , ( (pQmfImag[5] >> 1) + (pQmfImag[ 7] >> 1)));
+ t6 = fMultDiv2(p2_13_20[6] , pQmfImag[6] >> 1 );
+
+ mHybridImag[0] = (t1 + t3 + t5 + t6) << 2;
+ mHybridImag[1] = (- t1 - t3 - t5 + t6) << 2;
+}
+
+
+/*******************************************************************************
+ Functionname: eightChannelFiltering
+ *******************************************************************************
+
+ Description: fast 8-channel complex-valued filtering with 6-tap delay.
+
+ Arguments:
+
+ Return: none
+
+*******************************************************************************/
+/*!
+ 8 channel filter
+
+ Implementation using a FFT of length 8
+<pre>
+ prototype filter coefficients:
+ 0.00746082949812 0.02270420949825 0.04546865930473 0.07266113929591 0.09885108575264 0.11793710567217
+ 0.125
+ 0.11793710567217 0.09885108575264 0.07266113929591 0.04546865930473 0.02270420949825 0.00746082949812
+
+ Filter design:
+ N = 13; Q = 8;
+ h[q,n] = g[n] * exp(j * 2 * pi / Q * (q + .5) * (n - 6)); n = 0..(N-1), q = 0..(Q-1);
+
+ Time Signal: x[t];
+ Filter Bank Output
+ y[q,t] = conv(x[t],h[q,t]) = conv(h[q,t],x[t]) = sum(x[k] * h[q, t - k] ) = sum(h[q, k] * x[t - k] ); k = 0..(N-1);
+
+ y[q,t] = x[t - 12]*h[q, 12] + x[t - 11]*h[q, 11] + x[t - 10]*h[q, 10] + x[t - 9]*h[q, 9]
+ + x[t - 8]*h[q, 8] + x[t - 7]*h[q, 7]
+ + x[t - 6]*h[q, 6]
+ + x[t - 5]*h[q, 5] + x[t - 4]*h[q, 4]
+ + x[t - 3]*h[q, 3] + x[t - 2]*h[q, 2] + x[t - 1]*h[q, 1] + x[t - 0]*h[q, 0];
+
+ h'[q, n] = h[q,(N-1)-n] = g[n] * exp(j * 2 * pi / Q * (q + .5) * (6 - n)); n = 0..(N-1), q = 0..(Q-1);
+
+ y[q,t] = x[t - 12]*h'[q, 0] + x[t - 11]*h'[q, 1] + x[t - 10]*h'[q, 2] + x[t - 9]*h'[q, 3]
+ + x[t - 8]*h'[q, 4] + x[t - 7]*h'[q, 5]
+ + x[t - 6]*h'[q, 6]
+ + x[t - 5]*h'[q, 7] + x[t - 4]*h'[q, 8]
+ + x[t - 3]*h'[q, 9] + x[t - 2]*h'[q, 10] + x[t - 1]*h'[q, 11] + x[t - 0]*h'[q, 12];
+
+ Try to split off FFT Modulation Term:
+ FFT(x[t], q) = sum(x[t+k]*exp(-j*2*pi/N *q * k))
+ c m
+ Step 1: h'[q,n] = g[n] * ( exp(j * 2 * pi / 8 * .5 * (6 - n)) ) * ( exp (j * 2 * pi / 8 * q * (6 - n)) );
+
+ h'[q,n] = g[n] *c[n] * m[q,n]; (see above)
+ c[n] = exp( j * 2 * pi / 8 * .5 * (6 - n) );
+ m[q,n] = exp( j * 2 * pi / 8 * q * (6 - n) );
+
+ y[q,t] = x[t - 0]*g[0]*c[0]*m[q,0] + x[t - 1]*g[1]*c[ 1]*m[q, 1] + ...
+ ... + x[t - 12]*g[2]*c[12]*m[q,12];
+
+ |
+ n m *exp(-j*2*pi) | n' fft
+-------------------------------------------------------------------------------------------------------------------------
+ 0 exp( j * 2 * pi / 8 * q * 6) -> exp(-j * 2 * pi / 8 * q * 2) | 2 exp(-j * 2 * pi / 8 * q * 0)
+ 1 exp( j * 2 * pi / 8 * q * 5) -> exp(-j * 2 * pi / 8 * q * 3) | 3 exp(-j * 2 * pi / 8 * q * 1)
+ 2 exp( j * 2 * pi / 8 * q * 4) -> exp(-j * 2 * pi / 8 * q * 4) | 4 exp(-j * 2 * pi / 8 * q * 2)
+ 3 exp( j * 2 * pi / 8 * q * 3) -> exp(-j * 2 * pi / 8 * q * 5) | 5 exp(-j * 2 * pi / 8 * q * 3)
+ 4 exp( j * 2 * pi / 8 * q * 2) -> exp(-j * 2 * pi / 8 * q * 6) | 6 exp(-j * 2 * pi / 8 * q * 4)
+ 5 exp( j * 2 * pi / 8 * q * 1) -> exp(-j * 2 * pi / 8 * q * 7) | 7 exp(-j * 2 * pi / 8 * q * 5)
+ 6 exp( j * 2 * pi / 8 * q * 0) | 0 exp(-j * 2 * pi / 8 * q * 6)
+ 7 exp(-j * 2 * pi / 8 * q * 1) | 1 exp(-j * 2 * pi / 8 * q * 7)
+ 8 exp(-j * 2 * pi / 8 * q * 2) | 2
+ 9 exp(-j * 2 * pi / 8 * q * 3) | 3
+ 10 exp(-j * 2 * pi / 8 * q * 4) | 4
+ 11 exp(-j * 2 * pi / 8 * q * 5) | 5
+ 12 exp(-j * 2 * pi / 8 * q * 6) | 6
+
+
+ now use fft modulation coefficients
+ m[6] = = fft[0]
+ m[7] = = fft[1]
+ m[8] = m[ 0] = fft[2]
+ m[9] = m[ 1] = fft[3]
+ m[10] = m[ 2] = fft[4]
+ m[11] = m[ 3] = fft[5]
+ m[12] = m[ 4] = fft[6]
+ m[ 5] = fft[7]
+
+ y[q,t] = ( x[t- 6]*g[ 6]*c[ 6] ) * fft[q,0] +
+ ( x[t- 7]*g[ 7]*c[ 7] ) * fft[q,1] +
+ ( x[t- 0]*g[ 0]*c[ 0] + x[t- 8]*g[ 8]*c[ 8] ) * fft[q,2] +
+ ( x[t- 1]*g[ 1]*c[ 1] + x[t- 9]*g[ 9]*c[ 9] ) * fft[q,3] +
+ ( x[t- 2]*g[ 2]*c[ 2] + x[t-10]*g[10]*c[10] ) * fft[q,4] +
+ ( x[t- 3]*g[ 3]*c[ 3] + x[t-11]*g[11]*c[11] ) * fft[q,5] +
+ ( x[t- 4]*g[ 4]*c[ 4] + x[t-12]*g[12]*c[12] ) * fft[q,6] +
+ ( x[t- 5]*g[ 5]*c[ 5] ) * fft[q,7];
+
+ pre twiddle factors c[n] = exp(j * 2 * pi / 8 * .5 * (6 - n));
+ n c] | n c[n] | n c[n]
+---------------------------------------------------------------------------------------------------
+ 0 exp( j * 6 * pi / 8) | 1 exp( j * 5 * pi / 8) | 2 exp( j * 4 * pi / 8)
+ 3 exp( j * 3 * pi / 8) | 4 exp( j * 2 * pi / 8) | 5 exp( j * 1 * pi / 8)
+ 6 exp( j * 0 * pi / 8) | 7 exp(-j * 1 * pi / 8) | 8 exp(-j * 2 * pi / 8)
+ 9 exp(-j * 3 * pi / 8) | 10 exp(-j * 4 * pi / 8) | 11 exp(-j * 5 * pi / 8)
+ 12 exp(-j * 6 * pi / 8) | |
+</pre>
+*/
+
+/* defining rotation factors for *ChannelFiltering */
+
+#define cos0Pi FL2FXCONST_DBL( 1.f)
+#define sin0Pi FL2FXCONST_DBL( 0.f)
+
+#define cos1Pi FL2FXCONST_DBL(-1.f)
+#define sin1Pi FL2FXCONST_DBL( 0.f)
+
+#define cos1Pi_2 FL2FXCONST_DBL( 0.f)
+#define sin1Pi_2 FL2FXCONST_DBL( 1.f)
+
+#define cos1Pi_3 FL2FXCONST_DBL( 0.5f)
+#define sin1Pi_3 FL2FXCONST_DBL( 0.86602540378444f)
+
+#define cos0Pi_4 cos0Pi
+#define cos1Pi_4 FL2FXCONST_DBL(0.70710678118655f)
+#define cos2Pi_4 cos1Pi_2
+#define cos3Pi_4 (-cos1Pi_4)
+#define cos4Pi_4 (-cos0Pi_4)
+#define cos5Pi_4 cos3Pi_4
+#define cos6Pi_4 cos2Pi_4
+
+#define sin0Pi_4 sin0Pi
+#define sin1Pi_4 FL2FXCONST_DBL(0.70710678118655f)
+#define sin2Pi_4 sin1Pi_2
+#define sin3Pi_4 sin1Pi_4
+#define sin4Pi_4 sin0Pi_4
+#define sin5Pi_4 (-sin3Pi_4)
+#define sin6Pi_4 (-sin2Pi_4)
+
+#define cos0Pi_8 cos0Pi
+#define cos1Pi_8 FL2FXCONST_DBL(0.92387953251129f)
+#define cos2Pi_8 cos1Pi_4
+#define cos3Pi_8 FL2FXCONST_DBL(0.38268343236509f)
+#define cos4Pi_8 cos2Pi_4
+#define cos5Pi_8 (-cos3Pi_8)
+#define cos6Pi_8 (-cos2Pi_8)
+
+#define sin0Pi_8 sin0Pi
+#define sin1Pi_8 cos3Pi_8
+#define sin2Pi_8 sin1Pi_4
+#define sin3Pi_8 cos1Pi_8
+#define sin4Pi_8 sin2Pi_4
+#define sin5Pi_8 sin3Pi_8
+#define sin6Pi_8 sin1Pi_4
+
+#if defined(ARCH_PREFER_MULT_32x16)
+ #define FIXP_HYB FIXP_SGL
+ #define FIXP_CAST FX_DBL2FX_SGL
+#else
+ #define FIXP_HYB FIXP_DBL
+ #define FIXP_CAST
+#endif
+
+static const FIXP_HYB cr[13] =
+{
+ FIXP_CAST(cos6Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos4Pi_8),
+ FIXP_CAST(cos3Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos1Pi_8),
+ FIXP_CAST(cos0Pi_8),
+ FIXP_CAST(cos1Pi_8), FIXP_CAST(cos2Pi_8), FIXP_CAST(cos3Pi_8),
+ FIXP_CAST(cos4Pi_8), FIXP_CAST(cos5Pi_8), FIXP_CAST(cos6Pi_8)
+};
+
+static const FIXP_HYB ci[13] =
+{
+ FIXP_CAST( sin6Pi_8), FIXP_CAST( sin5Pi_8), FIXP_CAST( sin4Pi_8),
+ FIXP_CAST( sin3Pi_8), FIXP_CAST( sin2Pi_8), FIXP_CAST( sin1Pi_8),
+ FIXP_CAST( sin0Pi_8) ,
+ FIXP_CAST(-sin1Pi_8), FIXP_CAST(-sin2Pi_8), FIXP_CAST(-sin3Pi_8),
+ FIXP_CAST(-sin4Pi_8), FIXP_CAST(-sin5Pi_8), FIXP_CAST(-sin6Pi_8)
+};
+
+static void slotBasedEightChannelFiltering( const FIXP_DBL *pQmfReal,
+ const FIXP_DBL *pQmfImag,
+
+ FIXP_DBL *mHybridReal,
+ FIXP_DBL *mHybridImag)
+{
+
+ int bin;
+ FIXP_DBL _fft[128 + ALIGNMENT_DEFAULT - 1];
+ FIXP_DBL *fft = (FIXP_DBL *)ALIGN_PTR(_fft);
+
+#if defined(ARCH_PREFER_MULT_32x16)
+ const FIXP_SGL *p = p8_13_20; /* BASELINE_PS */
+#else
+ const FIXP_DBL *p = p8_13_20; /* BASELINE_PS */
+#endif
+
+ /* pre twiddeling */
+
+ /* x*(a*b + c*d) = fMultDiv2(x, fMultAddDiv2(fMultDiv2(a, b), c, d)) */
+ /* x*(a*b - c*d) = fMultDiv2(x, fMultSubDiv2(fMultDiv2(a, b), c, d)) */
+ FIXP_DBL accu1, accu2, accu3, accu4;
+
+ #define TWIDDLE_1(n_0,n_1,n_2) \
+ cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]); \
+ accu1 = fMultDiv2(p[n_0], accu1); \
+ accu2 = fMultDiv2(p[n_0], accu2); \
+ cplxMultDiv2(&accu3, &accu4, pQmfReal[n_1], pQmfImag[n_1], cr[n_1], ci[n_1]); \
+ accu3 = fMultDiv2(p[n_1], accu3); \
+ accu4 = fMultDiv2(p[n_1], accu4); \
+ fft[FIXP_FFT_IDX_R(n_2)] = accu1 + accu3; \
+ fft[FIXP_FFT_IDX_I(n_2)] = accu2 + accu4;
+
+ #define TWIDDLE_0(n_0,n_1) \
+ cplxMultDiv2(&accu1, &accu2, pQmfReal[n_0], pQmfImag[n_0], cr[n_0], ci[n_0]); \
+ fft[FIXP_FFT_IDX_R(n_1)] = fMultDiv2(p[n_0], accu1); \
+ fft[FIXP_FFT_IDX_I(n_1)] = fMultDiv2(p[n_0], accu2);
+
+ TWIDDLE_0( 6, 0)
+ TWIDDLE_0( 7, 1)
+
+ TWIDDLE_1( 0, 8, 2)
+ TWIDDLE_1( 1, 9, 3)
+ TWIDDLE_1( 2,10, 4)
+ TWIDDLE_1( 3,11, 5)
+ TWIDDLE_1( 4,12, 6)
+
+ TWIDDLE_0( 5, 7)
+
+ fft_8 (fft);
+
+ /* resort fft data into output array*/
+ for(bin=0; bin<8;bin++ ) {
+ mHybridReal[bin] = fft[FIXP_FFT_IDX_R(bin)] << 4;
+ mHybridImag[bin] = fft[FIXP_FFT_IDX_I(bin)] << 4;
+ }
+}
+
+
+/*******************************************************************************
+ Functionname: fillHybridDelayLine
+ *******************************************************************************
+
+ Description: The delay line of the hybrid filter is filled and copied from
+ left to right.
+
+ Return: none
+
+*******************************************************************************/
+
+void
+fillHybridDelayLine( FIXP_DBL **fixpQmfReal, /*!< Qmf real Values */
+ FIXP_DBL **fixpQmfImag, /*!< Qmf imag Values */
+ FIXP_DBL fixpHybridLeftR[12], /*!< Hybrid real Values left channel */
+ FIXP_DBL fixpHybridLeftI[12], /*!< Hybrid imag Values left channel */
+ FIXP_DBL fixpHybridRightR[12], /*!< Hybrid real Values right channel */
+ FIXP_DBL fixpHybridRightI[12], /*!< Hybrid imag Values right channel */
+ HANDLE_HYBRID hHybrid )
+{
+ int i;
+
+ for (i = 0; i < HYBRID_FILTER_DELAY; i++) {
+ slotBasedHybridAnalysis ( fixpQmfReal[i],
+ fixpQmfReal[i],
+ fixpHybridLeftR,
+ fixpHybridLeftI,
+ hHybrid );
+ }
+
+ FDKmemcpy(fixpHybridRightR, fixpHybridLeftR, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS);
+ FDKmemcpy(fixpHybridRightI, fixpHybridLeftI, sizeof(FIXP_DBL)*NO_SUB_QMF_CHANNELS);
+}
+
+
+/*******************************************************************************
+ Functionname: slotBasedHybridAnalysis
+ *******************************************************************************
+
+ Description: The lower QMF subbands are further split to provide better
+ frequency resolution for PS processing.
+
+ Return: none
+
+*******************************************************************************/
+
+
+void
+slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal, /*!< Qmf real Values */
+ FIXP_DBL *fixpQmfImag, /*!< Qmf imag Values */
+
+ FIXP_DBL fixpHybridReal[12], /*!< Hybrid real Values */
+ FIXP_DBL fixpHybridImag[12], /*!< Hybrid imag Values */
+
+ HANDLE_HYBRID hHybrid)
+{
+ int k, band;
+ HYBRID_RES hybridRes;
+ int chOffset = 0;
+
+ C_ALLOC_SCRATCH_START(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH);
+
+ FIXP_DBL *pTempImagSlot = pTempRealSlot + HYBRID_FILTER_LENGTH;
+ FIXP_DBL *pWorkRealSlot = pTempImagSlot + HYBRID_FILTER_LENGTH;
+ FIXP_DBL *pWorkImagSlot = pWorkRealSlot + HYBRID_FILTER_LENGTH;
+
+ /*!
+ Hybrid filtering is applied to the first hHybrid->nQmfBands QMF bands (3 when 10 or 20 stereo bands
+ are used, 5 when 34 stereo bands are used). For the remaining QMF bands a delay would be necessary.
+ But there is no need to implement a delay because there is a look-ahead of HYBRID_FILTER_DELAY = 6
+ QMF samples in the low-band buffer.
+ */
+
+ for(band = 0; band < hHybrid->nQmfBands; band++) {
+
+ /* get hybrid resolution per qmf band */
+ /* in case of baseline ps 10/20 band stereo mode : */
+ /* */
+ /* qmfBand[0] : 8 ( HYBRID_8_CPLX ) */
+ /* qmfBand[1] : 2 ( HYBRID_2_REAL ) */
+ /* qmfBand[2] : 2 ( HYBRID_2_REAL ) */
+ /* */
+ /* (split the 3 lower qmf band to 12 hybrid bands) */
+
+ hybridRes = (HYBRID_RES)hHybrid->pResolution[band];
+
+ FDKmemcpy(pWorkRealSlot, hHybrid->mQmfBufferRealSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+ FDKmemcpy(pWorkImagSlot, hHybrid->mQmfBufferImagSlot[band], hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+
+ pWorkRealSlot[hHybrid->qmfBufferMove] = fixpQmfReal[band];
+ pWorkImagSlot[hHybrid->qmfBufferMove] = fixpQmfImag[band];
+
+ FDKmemcpy(hHybrid->mQmfBufferRealSlot[band], pWorkRealSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+ FDKmemcpy(hHybrid->mQmfBufferImagSlot[band], pWorkImagSlot + 1, hHybrid->qmfBufferMove * sizeof(FIXP_DBL));
+
+ if (fixpQmfReal) {
+
+ /* actual filtering only if output signal requested */
+ switch( hybridRes ) {
+
+ /* HYBRID_2_REAL & HYBRID_8_CPLX are only needful for baseline ps */
+ case HYBRID_2_REAL:
+
+ slotBasedDualChannelFiltering( pWorkRealSlot,
+ pWorkImagSlot,
+ pTempRealSlot,
+ pTempImagSlot);
+ break;
+
+ case HYBRID_8_CPLX:
+
+ slotBasedEightChannelFiltering( pWorkRealSlot,
+ pWorkImagSlot,
+ pTempRealSlot,
+ pTempImagSlot);
+ break;
+
+ default:
+ FDK_ASSERT(0);
+ }
+
+ for(k = 0; k < (SCHAR)hybridRes; k++) {
+ fixpHybridReal [chOffset + k] = pTempRealSlot[k];
+ fixpHybridImag [chOffset + k] = pTempImagSlot[k];
+ }
+ chOffset += hybridRes;
+ } /* if (mHybridReal) */
+ }
+
+ /* group hybrid channels 3+4 -> 3 and 2+5 -> 2 */
+ fixpHybridReal[3] += fixpHybridReal[4];
+ fixpHybridImag[3] += fixpHybridImag[4];
+ fixpHybridReal[4] = (FIXP_DBL)0;
+ fixpHybridImag[4] = (FIXP_DBL)0;
+
+ fixpHybridReal[2] += fixpHybridReal[5];
+ fixpHybridImag[2] += fixpHybridImag[5];
+ fixpHybridReal[5] = (FIXP_DBL)0;
+ fixpHybridImag[5] = (FIXP_DBL)0;
+
+ /* free memory on scratch */
+ C_ALLOC_SCRATCH_END(pTempRealSlot, FIXP_DBL, 4*HYBRID_FILTER_LENGTH);
+
+}
+
+
+/*******************************************************************************
+ Functionname: slotBasedHybridSynthesis
+ *******************************************************************************
+
+ Description: The coefficients offering higher resolution for the lower QMF
+ channel are simply added prior to the synthesis with the 54
+ subbands QMF.
+
+ Arguments:
+
+ Return: none
+
+*******************************************************************************/
+
+/*! <pre>
+ l,r0(n) ---\
+ l,r1(n) ---- + --\
+ l,r2(n) ---/ \
+ + --> F0(w)
+ l,r3(n) ---\ /
+ l,r4(n) ---- + --/
+ l,r5(n) ---/
+
+
+ l,r6(n) ---\
+ + ---------> F1(w)
+ l,r7(n) ---/
+
+
+ l,r8(n) ---\
+ + ---------> F2(w)
+ l,r9(n) ---/
+
+ </pre>
+ Hybrid QMF synthesis filterbank for the 10 and 20 stereo-bands configurations. The
+ coefficients offering higher resolution for the lower QMF channel are simply added
+ prior to the synthesis with the 54 subbands QMF.
+
+ [see ISO/IEC 14496-3:2001/FDAM 2:2004(E) - Page 52]
+*/
+
+
+void
+slotBasedHybridSynthesis ( FIXP_DBL *fixpHybridReal, /*!< Hybrid real Values */
+ FIXP_DBL *fixpHybridImag, /*!< Hybrid imag Values */
+ FIXP_DBL *fixpQmfReal, /*!< Qmf real Values */
+ FIXP_DBL *fixpQmfImag, /*!< Qmf imag Values */
+ HANDLE_HYBRID hHybrid ) /*!< Handle to HYBRID struct. */
+{
+ int k, band;
+
+ HYBRID_RES hybridRes;
+ int chOffset = 0;
+
+ for(band = 0; band < hHybrid->nQmfBands; band++) {
+
+ FIXP_DBL qmfReal = FL2FXCONST_DBL(0.f);
+ FIXP_DBL qmfImag = FL2FXCONST_DBL(0.f);
+ hybridRes = (HYBRID_RES)hHybrid->pResolution[band];
+
+ for(k = 0; k < (SCHAR)hybridRes; k++) {
+ qmfReal += fixpHybridReal[chOffset + k];
+ qmfImag += fixpHybridImag[chOffset + k];
+ }
+
+ fixpQmfReal[band] = qmfReal;
+ fixpQmfImag[band] = qmfImag;
+
+ chOffset += hybridRes;
+ }
+}
+
+
+
diff --git a/libSBRdec/src/psdec_hybrid.h b/libSBRdec/src/psdec_hybrid.h
new file mode 100644
index 0000000..6503df9
--- /dev/null
+++ b/libSBRdec/src/psdec_hybrid.h
@@ -0,0 +1,165 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+#ifndef __HYBRID_H
+#define __HYBRID_H
+
+#include "sbrdecoder.h"
+
+
+#define HYBRID_FILTER_LENGTH 13
+#define HYBRID_FILTER_DELAY 6
+
+
+#define FAST_FILTER2
+#define FAST_FILTER4
+#define FAST_FILTER8
+#define FAST_FILTER12
+
+#define FFT_IDX_R(a) (2*a)
+#define FFT_IDX_I(a) (2*a+1)
+
+#define FIXP_FFT_IDX_R(a) (a<<1)
+#define FIXP_FFT_IDX_I(a) ((a<<1) + 1)
+
+
+typedef enum {
+
+ HYBRID_2_REAL = 2,
+ HYBRID_4_CPLX = 4,
+ HYBRID_8_CPLX = 8,
+ HYBRID_12_CPLX = 12
+
+} HYBRID_RES;
+
+typedef struct
+{
+ SCHAR nQmfBands;
+ SCHAR frameSize;
+ SCHAR qmfBufferMove;
+
+ UCHAR pResolution[3];
+
+ FIXP_DBL mQmfBufferRealSlot[3][HYBRID_FILTER_LENGTH]; /**< Stores old Qmf samples. */
+ FIXP_DBL mQmfBufferImagSlot[3][HYBRID_FILTER_LENGTH];
+ SCHAR sf_mQmfBuffer;
+
+} HYBRID;
+
+typedef HYBRID *HANDLE_HYBRID;
+
+void
+fillHybridDelayLine( FIXP_DBL **fixpQmfReal,
+ FIXP_DBL **fixpQmfImag,
+ FIXP_DBL fixpHybridLeftR[12],
+ FIXP_DBL fixpHybridLeftI[12],
+ FIXP_DBL fixpHybridRightR[12],
+ FIXP_DBL fixpHybridRightI[12],
+ HANDLE_HYBRID hHybrid );
+
+void
+slotBasedHybridAnalysis ( FIXP_DBL *fixpQmfReal,
+ FIXP_DBL *fixpQmfImag,
+
+ FIXP_DBL *fixpHybridReal,
+ FIXP_DBL *fixpHybridImag,
+
+ HANDLE_HYBRID hHybrid);
+
+
+void
+slotBasedHybridSynthesis ( FIXP_DBL *fixpHybridReal,
+ FIXP_DBL *fixpHybridImag,
+
+ FIXP_DBL *fixpQmfReal,
+ FIXP_DBL *fixpQmfImag,
+
+ HANDLE_HYBRID hHybrid );
+
+SBR_ERROR InitHybridFilterBank ( HANDLE_HYBRID hHybrid,
+ SCHAR frameSize,
+ SCHAR noBands,
+ const UCHAR *pResolution );
+
+
+#endif /* __HYBRID_H */
diff --git a/libSBRdec/src/sbr_crc.cpp b/libSBRdec/src/sbr_crc.cpp
new file mode 100644
index 0000000..760bd1f
--- /dev/null
+++ b/libSBRdec/src/sbr_crc.cpp
@@ -0,0 +1,183 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief CRC check coutines
+*/
+
+#include "sbr_crc.h"
+
+#include "FDK_bitstream.h"
+#include "transcendent.h"
+
+#define MAXCRCSTEP 16
+#define MAXCRCSTEP_LD 4
+
+/*!
+ \brief crc calculation
+*/
+static ULONG
+calcCRC (HANDLE_CRC hCrcBuf, ULONG bValue, int nBits)
+{
+ int i;
+ ULONG bMask = (1UL << (nBits - 1));
+
+ for (i = 0; i < nBits; i++, bMask >>= 1) {
+ USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0;
+ USHORT flag1 = (bMask & bValue) ? 1 : 0;
+
+ flag ^= flag1;
+ hCrcBuf->crcState <<= 1;
+ if (flag)
+ hCrcBuf->crcState ^= hCrcBuf->crcPoly;
+ }
+
+ return (hCrcBuf->crcState);
+}
+
+
+/*!
+ \brief crc
+*/
+static int
+getCrc (HANDLE_FDK_BITSTREAM hBs, ULONG NrBits)
+{
+ int i;
+ CRC_BUFFER CrcBuf;
+
+ CrcBuf.crcState = SBR_CRC_START;
+ CrcBuf.crcPoly = SBR_CRC_POLY;
+ CrcBuf.crcMask = SBR_CRC_MASK;
+
+ int CrcStep = NrBits>>MAXCRCSTEP_LD;
+
+ int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP);
+ ULONG bValue;
+
+ for (i = 0; i < CrcStep; i++) {
+ bValue = FDKreadBits (hBs, MAXCRCSTEP);
+ calcCRC (&CrcBuf, bValue, MAXCRCSTEP);
+ }
+
+ bValue = FDKreadBits (hBs, CrcNrBitsRest);
+ calcCRC (&CrcBuf, bValue, CrcNrBitsRest);
+
+ return (CrcBuf.crcState & SBR_CRC_RANGE);
+
+}
+
+
+/*!
+ \brief crc interface
+ \return 1: CRC OK, 0: CRC check failure
+*/
+int
+SbrCrcCheck (HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer */
+ LONG NrBits) /*!< max. CRC length */
+{
+ int crcResult = 1;
+ ULONG NrCrcBits;
+ ULONG crcCheckResult;
+ LONG NrBitsAvailable;
+ ULONG crcCheckSum;
+
+ crcCheckSum = FDKreadBits (hBs, 10);
+
+ NrBitsAvailable = FDKgetValidBits(hBs);
+ if (NrBitsAvailable <= 0){
+ return 0;
+ }
+
+ NrCrcBits = fixMin ((INT)NrBits, (INT)NrBitsAvailable);
+
+ crcCheckResult = getCrc (hBs, NrCrcBits);
+ FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs)) );
+
+
+ if (crcCheckResult != crcCheckSum) {
+ crcResult = 0;
+ }
+
+ return (crcResult);
+}
diff --git a/libSBRdec/src/sbr_crc.h b/libSBRdec/src/sbr_crc.h
new file mode 100644
index 0000000..542843d
--- /dev/null
+++ b/libSBRdec/src/sbr_crc.h
@@ -0,0 +1,123 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief CRC checking routines
+*/
+#ifndef __SBR_CRC_H
+#define __SBR_CRC_H
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+
+/* some useful crc polynoms:
+
+crc5: x^5+x^4+x^2+x^1+1
+crc6: x^6+x^5+x^3+x^2+x+1
+crc7: x^7+x^6+x^2+1
+crc8: x^8+x^2+x+x+1
+*/
+
+/* default SBR CRC */ /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */
+#define SBR_CRC_POLY 0x0233
+#define SBR_CRC_MASK 0x0200
+#define SBR_CRC_START 0x0000
+#define SBR_CRC_RANGE 0x03FF
+
+typedef struct
+{
+ USHORT crcState;
+ USHORT crcMask;
+ USHORT crcPoly;
+}
+CRC_BUFFER;
+
+typedef CRC_BUFFER *HANDLE_CRC;
+
+int SbrCrcCheck (HANDLE_FDK_BITSTREAM hBitBuf,
+ LONG NrCrcBits);
+
+
+#endif
diff --git a/libSBRdec/src/sbr_deb.cpp b/libSBRdec/src/sbr_deb.cpp
new file mode 100644
index 0000000..aa37ffe
--- /dev/null
+++ b/libSBRdec/src/sbr_deb.cpp
@@ -0,0 +1,90 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Print selected debug messages
+*/
+
+#include "sbr_deb.h"
+
diff --git a/libSBRdec/src/sbr_deb.h b/libSBRdec/src/sbr_deb.h
new file mode 100644
index 0000000..324dea9
--- /dev/null
+++ b/libSBRdec/src/sbr_deb.h
@@ -0,0 +1,94 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Debugging aids
+*/
+
+#ifndef __SBR_DEB_H
+#define __SBR_DEB_H
+
+#include "sbrdecoder.h"
+
+#endif
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
new file mode 100644
index 0000000..208120c
--- /dev/null
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -0,0 +1,1046 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Sbr decoder
+ This module provides the actual decoder implementation. The SBR data (side information) is already
+ decoded. Only three functions are provided:
+
+ \li 1.) createSbrDec(): One time initialization
+ \li 2.) resetSbrDec(): Called by sbr_Apply() when the information contained in an SBR_HEADER_ELEMENT requires a reset
+ and recalculation of important SBR structures.
+ \li 3.) sbr_dec(): The actual decoder. Calls the different tools such as filterbanks, lppTransposer(), and calculateSbrEnvelope()
+ [the envelope adjuster].
+
+ \sa sbr_dec(), \ref documentationOverview
+*/
+
+#include "sbr_dec.h"
+
+#include "sbr_ram.h"
+#include "env_extr.h"
+#include "env_calc.h"
+#include "scale.h"
+
+#include "genericStds.h"
+
+#include "sbrdec_drc.h"
+
+
+
+static void assignLcTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ FIXP_DBL **QmfBufferReal,
+ int noCols )
+{
+ int slot, i;
+ FIXP_DBL *ptr;
+
+ /* Number of QMF timeslots in the overlap buffer: */
+ ptr = hSbrDec->pSbrOverlapBuffer;
+ for(slot=0; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) {
+ QmfBufferReal[slot] = ptr; ptr += (64);
+ }
+
+ /* Assign timeslots to Workbuffer1 */
+ ptr = hSbrDec->WorkBuffer1;
+ for(i=0; i<noCols; i++) {
+ QmfBufferReal[slot] = ptr; ptr += (64);
+ slot++;
+ }
+}
+
+
+static void assignHqTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ FIXP_DBL **QmfBufferReal,
+ FIXP_DBL **QmfBufferImag,
+ int noCols )
+{
+ FIXP_DBL *ptr;
+ int slot;
+
+ /* Number of QMF timeslots in one half of a frame (size of Workbuffer1 or 2): */
+ int halflen = (noCols >> 1) + hSbrDec->LppTrans.pSettings->overlap;
+ int totCols = noCols + hSbrDec->LppTrans.pSettings->overlap;
+
+ /* Number of QMF timeslots in the overlap buffer: */
+ ptr = hSbrDec->pSbrOverlapBuffer;
+ for(slot=0; slot<hSbrDec->LppTrans.pSettings->overlap; slot++) {
+ QmfBufferReal[slot] = ptr; ptr += (64);
+ QmfBufferImag[slot] = ptr; ptr += (64);
+ }
+
+ /* Assign first half of timeslots to Workbuffer1 */
+ ptr = hSbrDec->WorkBuffer1;
+ for(; slot<halflen; slot++) {
+ QmfBufferReal[slot] = ptr; ptr += (64);
+ QmfBufferImag[slot] = ptr; ptr += (64);
+ }
+
+ /* Assign second half of timeslots to Workbuffer2 */
+ ptr = hSbrDec->WorkBuffer2;
+ for(; slot<totCols; slot++) {
+ QmfBufferReal[slot] = ptr; ptr += (64);
+ QmfBufferImag[slot] = ptr; ptr += (64);
+ }
+}
+
+
+static void assignTimeSlots( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ int noCols,
+ int useLP )
+{
+ /* assign qmf time slots */
+ hSbrDec->useLP = useLP;
+ if (useLP) {
+ hSbrDec->SynthesisQMF.flags |= QMF_FLAG_LP;
+ hSbrDec->AnalysiscQMF.flags |= QMF_FLAG_LP;
+ } else {
+ hSbrDec->SynthesisQMF.flags &= ~QMF_FLAG_LP;
+ hSbrDec->AnalysiscQMF.flags &= ~QMF_FLAG_LP;
+ }
+ if (!useLP)
+ assignHqTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, hSbrDec->QmfBufferImag, noCols );
+ else
+ {
+ assignLcTimeSlots( hSbrDec, hSbrDec->QmfBufferReal, noCols );
+ }
+}
+
+static void changeQmfType( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ int useLdTimeAlign )
+{
+ UINT synQmfFlags = hSbrDec->SynthesisQMF.flags;
+ UINT anaQmfFlags = hSbrDec->AnalysiscQMF.flags;
+ int resetSynQmf = 0;
+ int resetAnaQmf = 0;
+
+ /* assign qmf type */
+ if (useLdTimeAlign) {
+ if (synQmfFlags & QMF_FLAG_CLDFB) {
+ /* change the type to MPSLD */
+ synQmfFlags &= ~QMF_FLAG_CLDFB;
+ synQmfFlags |= QMF_FLAG_MPSLDFB;
+ resetSynQmf = 1;
+ }
+ if (anaQmfFlags & QMF_FLAG_CLDFB) {
+ /* change the type to MPSLD */
+ anaQmfFlags &= ~QMF_FLAG_CLDFB;
+ anaQmfFlags |= QMF_FLAG_MPSLDFB;
+ resetAnaQmf = 1;
+ }
+ } else {
+ if (synQmfFlags & QMF_FLAG_MPSLDFB) {
+ /* change the type to CLDFB */
+ synQmfFlags &= ~QMF_FLAG_MPSLDFB;
+ synQmfFlags |= QMF_FLAG_CLDFB;
+ resetSynQmf = 1;
+ }
+ if (anaQmfFlags & QMF_FLAG_MPSLDFB) {
+ /* change the type to CLDFB */
+ anaQmfFlags &= ~QMF_FLAG_MPSLDFB;
+ anaQmfFlags |= QMF_FLAG_CLDFB;
+ resetAnaQmf = 1;
+ }
+ }
+
+ if (resetAnaQmf) {
+ int qmfErr = qmfInitAnalysisFilterBank (
+ &hSbrDec->AnalysiscQMF,
+ hSbrDec->anaQmfStates,
+ hSbrDec->AnalysiscQMF.no_col,
+ hSbrDec->AnalysiscQMF.lsb,
+ hSbrDec->AnalysiscQMF.usb,
+ hSbrDec->AnalysiscQMF.no_channels,
+ anaQmfFlags | QMF_FLAG_KEEP_STATES
+ );
+ if (qmfErr != 0) {
+ FDK_ASSERT(0);
+ }
+ }
+
+ if (resetSynQmf) {
+ int qmfErr = qmfInitSynthesisFilterBank (
+ &hSbrDec->SynthesisQMF,
+ hSbrDec->pSynQmfStates,
+ hSbrDec->SynthesisQMF.no_col,
+ hSbrDec->SynthesisQMF.lsb,
+ hSbrDec->SynthesisQMF.usb,
+ hSbrDec->SynthesisQMF.no_channels,
+ synQmfFlags | QMF_FLAG_KEEP_STATES
+ );
+
+ if (qmfErr != 0) {
+ FDK_ASSERT(0);
+ }
+ }
+}
+
+
+/*!
+ \brief SBR decoder core function for one channel
+
+ \image html BufferMgmtDetailed-1632.png
+
+ Besides the filter states of the QMF filter bank and the LPC-states of
+ the LPP-Transposer, processing is mainly based on four buffers:
+ #timeIn, #timeOut, #WorkBuffer2 and #OverlapBuffer. The #WorkBuffer2
+ is reused for all channels and might be used by the core decoder, a
+ static overlap buffer is required for each channel. Du to in-place
+ processing, #timeIn and #timeOut point to identical locations.
+
+ The spectral data is organized in so-called slots, each slot
+ containing 64 bands of complex data. The number of slots per frame is
+ dependend on the frame size. For mp3PRO, there are 18 slots per frame
+ and 6 slots per #OverlapBuffer. It is not necessary to have the slots
+ in located consecutive address ranges.
+
+ To optimize memory usage and to minimize the number of memory
+ accesses, the memory management is organized as follows (Slot numbers
+ based on mp3PRO):
+
+ 1.) Input time domain signal is located in #timeIn, the last slots
+ (0..5) of the spectral data of the previous frame are located in the
+ #OverlapBuffer. In addition, #frameData of the current frame resides
+ in the upper part of #timeIn.
+
+ 2.) During the cplxAnalysisQmfFiltering(), 32 samples from #timeIn are transformed
+ into a slot of up to 32 complex spectral low band values at a
+ time. The first spectral slot -- nr. 6 -- is written at slot number
+ zero of #WorkBuffer2. #WorkBuffer2 will be completely filled with
+ spectral data.
+
+ 3.) LPP-Transposition in lppTransposer() is processed on 24 slots. During the
+ transposition, the high band part of the spectral data is replicated
+ based on the low band data.
+
+ Envelope Adjustment is processed on the high band part of the spectral
+ data only by calculateSbrEnvelope().
+
+ 4.) The cplxSynthesisQmfFiltering() creates 64 time domain samples out
+ of a slot of 64 complex spectral values at a time. The first 6 slots
+ in #timeOut are filled from the results of spectral slots 0..5 in the
+ #OverlapBuffer. The consecutive slots in timeOut are now filled with
+ the results of spectral slots 6..17.
+
+ 5.) The preprocessed slots 18..23 have to be stored in the
+ #OverlapBuffer.
+
+*/
+
+void
+sbr_dec ( HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ INT_PCM *timeIn, /*!< pointer to input time signal */
+ INT_PCM *timeOut, /*!< pointer to output time signal */
+ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */
+ INT_PCM *timeOutRight, /*!< pointer to output time signal */
+ const int strideIn, /*!< Time data traversal strideIn */
+ const int strideOut, /*!< Time data traversal strideOut */
+ HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */
+ const int applyProcessing, /*!< Flag for SBR operation */
+ HANDLE_PS_DEC h_ps_d,
+ const UINT flags
+ )
+{
+ int i, slot, reserve;
+ int saveLbScale;
+ int ov_len;
+ int lastSlotOffs;
+ FIXP_DBL maxVal;
+
+ /* 1+1/3 frames of spectral data: */
+ FIXP_DBL **QmfBufferReal = hSbrDec->QmfBufferReal;
+ FIXP_DBL **QmfBufferImag = hSbrDec->QmfBufferImag;
+
+ /* Number of QMF timeslots in the overlap buffer: */
+ ov_len = hSbrDec->LppTrans.pSettings->overlap;
+
+ /* Number of QMF slots per frame */
+ int noCols = hHeaderData->numberTimeSlots * hHeaderData->timeStep;
+
+ /* assign qmf time slots */
+ if ( ((flags & SBRDEC_LOW_POWER ) ? 1 : 0) != ((hSbrDec->SynthesisQMF.flags & QMF_FLAG_LP) ? 1 : 0) ) {
+ assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, flags & SBRDEC_LOW_POWER);
+ }
+
+ if (flags & SBRDEC_ELD_GRID) {
+ /* Choose the right low delay filter bank */
+ changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 );
+ }
+
+ /*
+ low band codec signal subband filtering
+ */
+
+ {
+ C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64));
+
+ qmfAnalysisFiltering( &hSbrDec->AnalysiscQMF,
+ QmfBufferReal + ov_len,
+ QmfBufferImag + ov_len,
+ &hSbrDec->sbrScaleFactor,
+ timeIn,
+ strideIn,
+ qmfTemp
+ );
+
+ C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64));
+ }
+
+ /*
+ Clear upper half of spectrum
+ */
+ {
+ int nAnalysisBands = hHeaderData->numberOfAnalysisBands;
+
+ if (! (flags & SBRDEC_LOW_POWER)) {
+ for (slot = ov_len; slot < noCols+ov_len; slot++) {
+ FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+ FDKmemclear(&QmfBufferImag[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+ }
+ } else
+ for (slot = ov_len; slot < noCols+ov_len; slot++) {
+ FDKmemclear(&QmfBufferReal[slot][nAnalysisBands],((64)-nAnalysisBands)*sizeof(FIXP_DBL));
+ }
+ }
+
+
+
+ /*
+ Shift spectral data left to gain accuracy in transposer and adjustor
+ */
+ maxVal = maxSubbandSample( QmfBufferReal,
+ (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+ 0,
+ hSbrDec->AnalysiscQMF.lsb,
+ ov_len,
+ noCols+ov_len );
+
+ reserve = fixMax(0,CntLeadingZeros(maxVal)-1) ;
+ reserve = fixMin(reserve,DFRACT_BITS-1-hSbrDec->sbrScaleFactor.lb_scale);
+
+ /* If all data is zero, lb_scale could become too large */
+ rescaleSubbandSamples( QmfBufferReal,
+ (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+ 0,
+ hSbrDec->AnalysiscQMF.lsb,
+ ov_len,
+ noCols+ov_len,
+ reserve);
+
+ hSbrDec->sbrScaleFactor.lb_scale += reserve;
+
+ /*
+ save low band scale, wavecoding or parametric stereo may modify it
+ */
+ saveLbScale = hSbrDec->sbrScaleFactor.lb_scale;
+
+
+ if (applyProcessing)
+ {
+ UCHAR * borders = hFrameData->frameInfo.borders;
+ lastSlotOffs = borders[hFrameData->frameInfo.nEnvelopes] - hHeaderData->numberTimeSlots;
+
+ FIXP_DBL degreeAlias[(64)];
+
+ /* The transposer will override most values in degreeAlias[].
+ The array needs to be cleared at least from lowSubband to highSubband before. */
+ if (flags & SBRDEC_LOW_POWER)
+ FDKmemclear(&degreeAlias[hHeaderData->freqBandData.lowSubband], (hHeaderData->freqBandData.highSubband-hHeaderData->freqBandData.lowSubband)*sizeof(FIXP_DBL));
+
+ /*
+ Inverse filtering of lowband and transposition into the SBR-frequency range
+ */
+
+ lppTransposer ( &hSbrDec->LppTrans,
+ &hSbrDec->sbrScaleFactor,
+ QmfBufferReal,
+ degreeAlias, // only used if useLP = 1
+ QmfBufferImag,
+ flags & SBRDEC_LOW_POWER,
+ hHeaderData->timeStep,
+ borders[0],
+ lastSlotOffs,
+ hHeaderData->freqBandData.nInvfBands,
+ hFrameData->sbr_invf_mode,
+ hPrevFrameData->sbr_invf_mode );
+
+
+
+
+
+ /*
+ Adjust envelope of current frame.
+ */
+
+ calculateSbrEnvelope (&hSbrDec->sbrScaleFactor,
+ &hSbrDec->SbrCalculateEnvelope,
+ hHeaderData,
+ hFrameData,
+ QmfBufferReal,
+ QmfBufferImag,
+ flags & SBRDEC_LOW_POWER,
+
+ degreeAlias,
+ flags,
+ (hHeaderData->frameErrorFlag || hPrevFrameData->frameErrorFlag));
+
+
+ /*
+ Update hPrevFrameData (to be used in the next frame)
+ */
+ for (i=0; i<hHeaderData->freqBandData.nInvfBands; i++) {
+ hPrevFrameData->sbr_invf_mode[i] = hFrameData->sbr_invf_mode[i];
+ }
+ hPrevFrameData->coupling = hFrameData->coupling;
+ hPrevFrameData->stopPos = borders[hFrameData->frameInfo.nEnvelopes];
+ hPrevFrameData->ampRes = hFrameData->ampResolutionCurrentFrame;
+ }
+ else {
+ /* Reset hb_scale if no highband is present, because hb_scale is considered in the QMF-synthesis */
+ hSbrDec->sbrScaleFactor.hb_scale = saveLbScale;
+ }
+
+
+ for (i=0; i<LPC_ORDER; i++){
+ /*
+ Store the unmodified qmf Slots values (required for LPC filtering)
+ */
+ if (! (flags & SBRDEC_LOW_POWER)) {
+ FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+ FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesImag[i], QmfBufferImag[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+ } else
+ FDKmemcpy(hSbrDec->LppTrans.lpcFilterStatesReal[i], QmfBufferReal[noCols-LPC_ORDER+i], hSbrDec->AnalysiscQMF.lsb*sizeof(FIXP_DBL));
+ }
+
+ /*
+ Synthesis subband filtering.
+ */
+
+ if ( ! (flags & SBRDEC_PS_DECODED) ) {
+
+ {
+ int outScalefactor = 0;
+
+ if (h_ps_d != NULL) {
+ h_ps_d->procFrameBased = 1; /* we here do frame based processing */
+ }
+
+
+ sbrDecoder_drcApply(&hSbrDec->sbrDrcChannel,
+ QmfBufferReal,
+ (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+ hSbrDec->SynthesisQMF.no_col,
+ &outScalefactor
+ );
+
+
+
+ qmfChangeOutScalefactor(&hSbrDec->SynthesisQMF, outScalefactor );
+
+ {
+ C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2*(64));
+
+ qmfSynthesisFiltering( &hSbrDec->SynthesisQMF,
+ QmfBufferReal,
+ (flags & SBRDEC_LOW_POWER) ? NULL : QmfBufferImag,
+ &hSbrDec->sbrScaleFactor,
+ hSbrDec->LppTrans.pSettings->overlap,
+ timeOut,
+ strideOut,
+ qmfTemp);
+
+ C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2*(64));
+ }
+
+ }
+
+ } else { /* (flags & SBRDEC_PS_DECODED) */
+ INT i, sdiff, outScalefactor, scaleFactorLowBand, scaleFactorHighBand;
+ SCHAR scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;
+
+ HANDLE_QMF_FILTER_BANK synQmf = &hSbrDec->SynthesisQMF;
+ HANDLE_QMF_FILTER_BANK synQmfRight = &hSbrDecRight->SynthesisQMF;
+
+ /* adapt scaling */
+ sdiff = hSbrDec->sbrScaleFactor.lb_scale - reserve; /* Scaling difference */
+ scaleFactorHighBand = sdiff - hSbrDec->sbrScaleFactor.hb_scale; /* Scale of current high band */
+ scaleFactorLowBand_ov = sdiff - hSbrDec->sbrScaleFactor.ov_lb_scale; /* Scale of low band overlapping QMF data */
+ scaleFactorLowBand_no_ov = sdiff - hSbrDec->sbrScaleFactor.lb_scale; /* Scale of low band current QMF data */
+ outScalefactor = 0; /* Initial output scale */
+
+ if (h_ps_d->procFrameBased == 1) /* If we have switched from frame to slot based processing copy filter states */
+ { /* procFrameBased will be unset later */
+ /* copy filter states from left to right */
+ FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates, ((640)-(64))*sizeof(FIXP_QSS));
+ }
+
+ /* scale ALL qmf vales ( real and imag ) of mono / left channel to the
+ same scale factor ( ov_lb_sf, lb_sf and hq_sf ) */
+ scalFilterBankValues( h_ps_d, /* parametric stereo decoder handle */
+ QmfBufferReal, /* qmf filterbank values */
+ QmfBufferImag, /* qmf filterbank values */
+ synQmf->lsb, /* sbr start subband */
+ hSbrDec->sbrScaleFactor.ov_lb_scale,
+ hSbrDec->sbrScaleFactor.lb_scale,
+ &scaleFactorLowBand_ov, /* adapt scaling values */
+ &scaleFactorLowBand_no_ov, /* adapt scaling values */
+ hSbrDec->sbrScaleFactor.hb_scale, /* current frame ( highband ) */
+ &scaleFactorHighBand,
+ synQmf->no_col);
+
+ /* use the same synthese qmf values for left and right channel */
+ synQmfRight->no_col = synQmf->no_col;
+ synQmfRight->lsb = synQmf->lsb;
+ synQmfRight->usb = synQmf->usb;
+
+ int env=0;
+
+ outScalefactor += (SCAL_HEADROOM+1); /* psDiffScale! */
+
+ {
+ C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, 2*(64));
+
+ int maxShift = 0;
+
+ if (hSbrDec->sbrDrcChannel.prevFact_exp > maxShift) {
+ maxShift = hSbrDec->sbrDrcChannel.prevFact_exp;
+ }
+ if (hSbrDec->sbrDrcChannel.currFact_exp > maxShift) {
+ maxShift = hSbrDec->sbrDrcChannel.currFact_exp;
+ }
+ if (hSbrDec->sbrDrcChannel.nextFact_exp > maxShift) {
+ maxShift = hSbrDec->sbrDrcChannel.nextFact_exp;
+ }
+
+ /* copy DRC data to right channel (with PS both channels use the same DRC gains) */
+ FDKmemcpy(&hSbrDecRight->sbrDrcChannel, &hSbrDec->sbrDrcChannel, sizeof(SBRDEC_DRC_CHANNEL));
+
+ for (i = 0; i < synQmf->no_col; i++) { /* ----- no_col loop ----- */
+
+ INT outScalefactorR, outScalefactorL;
+ outScalefactorR = outScalefactorL = outScalefactor;
+
+ /* qmf timeslot of right channel */
+ FIXP_DBL* rQmfReal = pWorkBuffer;
+ FIXP_DBL* rQmfImag = pWorkBuffer + 64;
+
+
+ {
+ if ( i == h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env] ) {
+ initSlotBasedRotation( h_ps_d, env, hHeaderData->freqBandData.highSubband );
+ env++;
+ }
+
+ ApplyPsSlot( h_ps_d, /* parametric stereo decoder handle */
+ (QmfBufferReal + i), /* one timeslot of left/mono channel */
+ (QmfBufferImag + i), /* one timeslot of left/mono channel */
+ rQmfReal, /* one timeslot or right channel */
+ rQmfImag); /* one timeslot or right channel */
+ }
+
+
+ scaleFactorLowBand = (i<(6)) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov;
+
+
+ sbrDecoder_drcApplySlot ( /* right channel */
+ &hSbrDecRight->sbrDrcChannel,
+ rQmfReal,
+ rQmfImag,
+ i,
+ synQmfRight->no_col,
+ maxShift
+ );
+
+ outScalefactorR += maxShift;
+
+ sbrDecoder_drcApplySlot ( /* left channel */
+ &hSbrDec->sbrDrcChannel,
+ *(QmfBufferReal + i),
+ *(QmfBufferImag + i),
+ i,
+ synQmf->no_col,
+ maxShift
+ );
+
+ outScalefactorL += maxShift;
+
+
+ /* scale filter states for left and right channel */
+ qmfChangeOutScalefactor( synQmf, outScalefactorL );
+ qmfChangeOutScalefactor( synQmfRight, outScalefactorR );
+
+ {
+
+ qmfSynthesisFilteringSlot( synQmfRight,
+ rQmfReal, /* QMF real buffer */
+ rQmfImag, /* QMF imag buffer */
+ scaleFactorLowBand,
+ scaleFactorHighBand,
+ timeOutRight+(i*synQmf->no_channels*strideOut),
+ strideOut,
+ pWorkBuffer);
+
+ qmfSynthesisFilteringSlot( synQmf,
+ *(QmfBufferReal + i), /* QMF real buffer */
+ *(QmfBufferImag + i), /* QMF imag buffer */
+ scaleFactorLowBand,
+ scaleFactorHighBand,
+ timeOut+(i*synQmf->no_channels*strideOut),
+ strideOut,
+ pWorkBuffer);
+
+ }
+ } /* no_col loop i */
+
+ /* scale back (6) timeslots look ahead for hybrid filterbank to original value */
+ rescalFilterBankValues( h_ps_d,
+ QmfBufferReal,
+ QmfBufferImag,
+ synQmf->lsb,
+ synQmf->no_col );
+
+ C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, 2*(64));
+ }
+ }
+
+ sbrDecoder_drcUpdateChannel( &hSbrDec->sbrDrcChannel );
+
+
+ /*
+ Update overlap buffer
+ Even bands above usb are copied to avoid outdated spectral data in case
+ the stop frequency raises.
+ */
+
+ if (hSbrDec->LppTrans.pSettings->overlap > 0)
+ {
+ if (! (flags & SBRDEC_LOW_POWER)) {
+ for ( i=0; i<hSbrDec->LppTrans.pSettings->overlap; i++ ) {
+ FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL));
+ FDKmemcpy(QmfBufferImag[i], QmfBufferImag[i+noCols], (64)*sizeof(FIXP_DBL));
+ }
+ } else
+ for ( i=0; i<hSbrDec->LppTrans.pSettings->overlap; i++ ) {
+ FDKmemcpy(QmfBufferReal[i], QmfBufferReal[i+noCols], (64)*sizeof(FIXP_DBL));
+ }
+ }
+
+ hSbrDec->sbrScaleFactor.ov_lb_scale = saveLbScale;
+
+ /* Save current frame status */
+ hPrevFrameData->frameErrorFlag = hHeaderData->frameErrorFlag;
+
+} // sbr_dec()
+
+
+/*!
+ \brief Creates sbr decoder structure
+ \return errorCode, 0 if successful
+*/
+SBR_ERROR
+createSbrDec (SBR_CHANNEL * hSbrChannel,
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
+ TRANSPOSER_SETTINGS *pSettings,
+ const int downsampleFac, /*!< Downsampling factor */
+ const UINT qmfFlags, /*!< flags -> 1: HQ/LP selector, 2: CLDFB */
+ const UINT flags,
+ const int overlap,
+ int chan) /*!< Channel for which to assign buffers etc. */
+
+{
+ SBR_ERROR err = SBRDEC_OK;
+ int timeSlots = hHeaderData->numberTimeSlots; /* Number of SBR slots per frame */
+ int noCols = timeSlots * hHeaderData->timeStep; /* Number of QMF slots per frame */
+ HANDLE_SBR_DEC hs = &(hSbrChannel->SbrDec);
+
+ /* Initialize scale factors */
+ hs->sbrScaleFactor.ov_lb_scale = 0;
+ hs->sbrScaleFactor.ov_hb_scale = 0;
+ hs->sbrScaleFactor.hb_scale = 0;
+
+
+ /*
+ create envelope calculator
+ */
+ err = createSbrEnvelopeCalc (&hs->SbrCalculateEnvelope,
+ hHeaderData,
+ chan,
+ flags);
+ if (err != SBRDEC_OK) {
+ return err;
+ }
+
+ /*
+ create QMF filter banks
+ */
+ {
+ int qmfErr;
+
+ qmfErr = qmfInitAnalysisFilterBank (
+ &hs->AnalysiscQMF,
+ hs->anaQmfStates,
+ noCols,
+ hHeaderData->freqBandData.lowSubband,
+ hHeaderData->freqBandData.highSubband,
+ hHeaderData->numberOfAnalysisBands,
+ qmfFlags & (~QMF_FLAG_KEEP_STATES)
+ );
+ if (qmfErr != 0) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+ }
+ if (hs->pSynQmfStates == NULL) {
+ hs->pSynQmfStates = GetRam_sbr_QmfStatesSynthesis(chan);
+ if (hs->pSynQmfStates == NULL)
+ return SBRDEC_MEM_ALLOC_FAILED;
+ }
+
+ {
+ int qmfErr;
+
+ qmfErr = qmfInitSynthesisFilterBank (
+ &hs->SynthesisQMF,
+ hs->pSynQmfStates,
+ noCols,
+ hHeaderData->freqBandData.lowSubband,
+ hHeaderData->freqBandData.highSubband,
+ (64) / downsampleFac,
+ qmfFlags & (~QMF_FLAG_KEEP_STATES)
+ );
+
+ if (qmfErr != 0) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+ }
+ initSbrPrevFrameData (&hSbrChannel->prevFrameData, timeSlots);
+
+ /*
+ create transposer
+ */
+ err = createLppTransposer (&hs->LppTrans,
+ pSettings,
+ hHeaderData->freqBandData.lowSubband,
+ hHeaderData->freqBandData.v_k_master,
+ hHeaderData->freqBandData.numMaster,
+ hs->SynthesisQMF.usb,
+ timeSlots,
+ hs->AnalysiscQMF.no_col,
+ hHeaderData->freqBandData.freqBandTableNoise,
+ hHeaderData->freqBandData.nNfb,
+ hHeaderData->sbrProcSmplRate,
+ chan,
+ overlap );
+ if (err != SBRDEC_OK) {
+ return err;
+ }
+
+ /* The CLDFB does not have overlap */
+ if ((qmfFlags & QMF_FLAG_CLDFB) == 0) {
+ if (hs->pSbrOverlapBuffer == NULL) {
+ hs->pSbrOverlapBuffer = GetRam_sbr_OverlapBuffer(chan);
+ if (hs->pSbrOverlapBuffer == NULL) {
+ return SBRDEC_MEM_ALLOC_FAILED;
+ }
+ } else {
+ /* Clear overlap buffer */
+ FDKmemclear( hs->pSbrOverlapBuffer,
+ sizeof(FIXP_DBL) * 2 * (6) * (64)
+ );
+ }
+ }
+
+ /* assign qmf time slots */
+ assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP);
+
+ return err;
+}
+
+/*!
+ \brief Delete sbr decoder structure
+ \return errorCode, 0 if successful
+*/
+int
+deleteSbrDec (SBR_CHANNEL * hSbrChannel)
+{
+ HANDLE_SBR_DEC hs = &hSbrChannel->SbrDec;
+
+ deleteSbrEnvelopeCalc (&hs->SbrCalculateEnvelope);
+
+ /* delete QMF filter states */
+ if (hs->pSynQmfStates != NULL) {
+ FreeRam_sbr_QmfStatesSynthesis(&hs->pSynQmfStates);
+ }
+
+
+ if (hs->pSbrOverlapBuffer != NULL) {
+ FreeRam_sbr_OverlapBuffer(&hs->pSbrOverlapBuffer);
+ }
+
+ return 0;
+}
+
+
+/*!
+ \brief resets sbr decoder structure
+ \return errorCode, 0 if successful
+*/
+SBR_ERROR
+resetSbrDec (HANDLE_SBR_DEC hSbrDec,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+ const int useLP,
+ const int downsampleFac
+ )
+{
+ SBR_ERROR sbrError = SBRDEC_OK;
+
+ int old_lsb = hSbrDec->SynthesisQMF.lsb;
+ int new_lsb = hHeaderData->freqBandData.lowSubband;
+ int l, startBand, stopBand, startSlot, size;
+
+ int source_scale, target_scale, delta_scale, target_lsb, target_usb, reserve;
+ FIXP_DBL maxVal;
+
+ /* overlapBuffer point to first (6) slots */
+ FIXP_DBL **OverlapBufferReal = hSbrDec->QmfBufferReal;
+ FIXP_DBL **OverlapBufferImag = hSbrDec->QmfBufferImag;
+
+ /* assign qmf time slots */
+ assignTimeSlots( hSbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, useLP);
+
+
+
+ resetSbrEnvelopeCalc (&hSbrDec->SbrCalculateEnvelope);
+
+ hSbrDec->SynthesisQMF.lsb = hHeaderData->freqBandData.lowSubband;
+ hSbrDec->SynthesisQMF.usb = fixMin((INT)hSbrDec->SynthesisQMF.no_channels, (INT)hHeaderData->freqBandData.highSubband);
+
+ hSbrDec->AnalysiscQMF.lsb = hSbrDec->SynthesisQMF.lsb;
+ hSbrDec->AnalysiscQMF.usb = hSbrDec->SynthesisQMF.usb;
+
+
+ /*
+ The following initialization of spectral data in the overlap buffer
+ is required for dynamic x-over or a change of the start-freq for 2 reasons:
+
+ 1. If the lowband gets _wider_, unadjusted data would remain
+
+ 2. If the lowband becomes _smaller_, the highest bands of the old lowband
+ must be cleared because the whitening would be affected
+ */
+ startBand = old_lsb;
+ stopBand = new_lsb;
+ startSlot = hHeaderData->timeStep * (hPrevFrameData->stopPos - hHeaderData->numberTimeSlots);
+ size = fixMax(0,stopBand-startBand);
+
+ /* keep already adjusted data in the x-over-area */
+ if (!useLP) {
+ for (l=startSlot; l<hSbrDec->LppTrans.pSettings->overlap; l++) {
+ FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL));
+ FDKmemclear(&OverlapBufferImag[l][startBand], size*sizeof(FIXP_DBL));
+ }
+ } else
+ for (l=startSlot; l<hSbrDec->LppTrans.pSettings->overlap ; l++) {
+ FDKmemclear(&OverlapBufferReal[l][startBand], size*sizeof(FIXP_DBL));
+ }
+
+
+ /*
+ reset LPC filter states
+ */
+ startBand = fixMin(old_lsb,new_lsb);
+ stopBand = fixMax(old_lsb,new_lsb);
+ size = fixMax(0,stopBand-startBand);
+
+ FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[0][startBand], size*sizeof(FIXP_DBL));
+ FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesReal[1][startBand], size*sizeof(FIXP_DBL));
+ if (!useLP) {
+ FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[0][startBand], size*sizeof(FIXP_DBL));
+ FDKmemclear(&hSbrDec->LppTrans.lpcFilterStatesImag[1][startBand], size*sizeof(FIXP_DBL));
+ }
+
+
+ /*
+ Rescale already processed spectral data between old and new x-over frequency.
+ This must be done because of the separate scalefactors for lowband and highband.
+ */
+ startBand = fixMin(old_lsb,new_lsb);
+ stopBand = fixMax(old_lsb,new_lsb);
+
+ if (new_lsb > old_lsb) {
+ /* The x-over-area was part of the highband before and will now belong to the lowband */
+ source_scale = hSbrDec->sbrScaleFactor.ov_hb_scale;
+ target_scale = hSbrDec->sbrScaleFactor.ov_lb_scale;
+ target_lsb = 0;
+ target_usb = old_lsb;
+ }
+ else {
+ /* The x-over-area was part of the lowband before and will now belong to the highband */
+ source_scale = hSbrDec->sbrScaleFactor.ov_lb_scale;
+ target_scale = hSbrDec->sbrScaleFactor.ov_hb_scale;
+ /* jdr: The values old_lsb and old_usb might be wrong because the previous frame might have been "upsamling". */
+ target_lsb = hSbrDec->SynthesisQMF.lsb;
+ target_usb = hSbrDec->SynthesisQMF.usb;
+ }
+
+ /* Shift left all samples of the x-over-area as much as possible
+ An unnecessary coarse scale could cause ov_lb_scale or ov_hb_scale to be
+ adapted and the accuracy in the next frame would seriously suffer! */
+
+ maxVal = maxSubbandSample( OverlapBufferReal,
+ (useLP) ? NULL : OverlapBufferImag,
+ startBand,
+ stopBand,
+ 0,
+ startSlot);
+
+ reserve = CntLeadingZeros(maxVal)-1;
+ reserve = fixMin(reserve,DFRACT_BITS-1-source_scale);
+
+ rescaleSubbandSamples( OverlapBufferReal,
+ (useLP) ? NULL : OverlapBufferImag,
+ startBand,
+ stopBand,
+ 0,
+ startSlot,
+ reserve);
+ source_scale += reserve;
+
+ delta_scale = target_scale - source_scale;
+
+ if (delta_scale > 0) { /* x-over-area is dominant */
+ delta_scale = -delta_scale;
+ startBand = target_lsb;
+ stopBand = target_usb;
+
+ if (new_lsb > old_lsb) {
+ /* The lowband has to be rescaled */
+ hSbrDec->sbrScaleFactor.ov_lb_scale = source_scale;
+ }
+ else {
+ /* The highband has be be rescaled */
+ hSbrDec->sbrScaleFactor.ov_hb_scale = source_scale;
+ }
+ }
+
+ FDK_ASSERT(startBand <= stopBand);
+
+ if (!useLP) {
+ for (l=0; l<startSlot; l++) {
+ scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale );
+ scaleValues( OverlapBufferImag[l] + startBand, stopBand-startBand, delta_scale );
+ }
+ } else
+ for (l=0; l<startSlot; l++) {
+ scaleValues( OverlapBufferReal[l] + startBand, stopBand-startBand, delta_scale );
+ }
+
+
+ /*
+ Initialize transposer and limiter
+ */
+ sbrError = resetLppTransposer (&hSbrDec->LppTrans,
+ hHeaderData->freqBandData.lowSubband,
+ hHeaderData->freqBandData.v_k_master,
+ hHeaderData->freqBandData.numMaster,
+ hHeaderData->freqBandData.freqBandTableNoise,
+ hHeaderData->freqBandData.nNfb,
+ hHeaderData->freqBandData.highSubband,
+ hHeaderData->sbrProcSmplRate);
+ if (sbrError != SBRDEC_OK)
+ return sbrError;
+
+ sbrError = ResetLimiterBands ( hHeaderData->freqBandData.limiterBandTable,
+ &hHeaderData->freqBandData.noLimiterBands,
+ hHeaderData->freqBandData.freqBandTable[0],
+ hHeaderData->freqBandData.nSfb[0],
+ hSbrDec->LppTrans.pSettings->patchParam,
+ hSbrDec->LppTrans.pSettings->noOfPatches,
+ hHeaderData->bs_data.limiterBands);
+
+
+ return sbrError;
+}
diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h
new file mode 100644
index 0000000..309327f
--- /dev/null
+++ b/libSBRdec/src/sbr_dec.h
@@ -0,0 +1,210 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Sbr decoder
+*/
+#ifndef __SBR_DEC_H
+#define __SBR_DEC_H
+
+#include "sbrdecoder.h"
+
+#include "lpp_tran.h"
+#include "qmf.h"
+#include "env_calc.h"
+#include "FDK_audio.h"
+
+
+#include "sbrdec_drc.h"
+
+#define SACDEC_ALIGNMENT_FIX
+
+typedef struct
+{
+ QMF_FILTER_BANK AnalysiscQMF;
+ QMF_FILTER_BANK SynthesisQMF;
+
+ SBR_CALCULATE_ENVELOPE SbrCalculateEnvelope;
+ SBR_LPP_TRANS LppTrans;
+
+ QMF_SCALE_FACTOR sbrScaleFactor;
+ QMF_SCALE_FACTOR sbrScaleFactorRight;
+
+ /*! Delayed spectral data needed for the dynamic framing of SBR. Not required in case of CLDFB */
+ FIXP_DBL * pSbrOverlapBuffer;
+
+ /* References to workbuffers */
+ FIXP_DBL * WorkBuffer1;
+ FIXP_DBL * WorkBuffer2;
+
+ /* QMF filter states */
+ FIXP_QAS anaQmfStates[(320)];
+ FIXP_QSS * pSynQmfStates;
+
+ /* Reference pointer arrays for QMF time slots,
+ mixed among overlap and current slots. */
+ FIXP_DBL * QmfBufferReal[(((1024)/(32))+(6))];
+ FIXP_DBL * QmfBufferImag[(((1024)/(32))+(6))];
+ int useLP;
+
+ /* QMF domain extension time slot reference pointer array */
+
+ SBRDEC_DRC_CHANNEL sbrDrcChannel;
+
+} SBR_DEC;
+
+typedef SBR_DEC *HANDLE_SBR_DEC;
+
+
+typedef struct
+{
+ SBR_FRAME_DATA frameData[(1)+1];
+ SBR_PREV_FRAME_DATA prevFrameData;
+ SBR_DEC SbrDec;
+}
+SBR_CHANNEL;
+
+typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL;
+
+void
+SbrDecodeAndProcess (HANDLE_SBR_DEC hSbrDec,
+ INT_PCM *timeIn,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_FRAME_DATA hFrameData,
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+ int applyProcessing,
+ int channelNr
+ , UCHAR useLP
+ );
+
+
+void
+SbrConstructTimeOutput (HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ INT_PCM *timeOut, /*!< pointer to output time signal */
+ HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */
+ int channelNr
+ ,UCHAR useLP
+ );
+
+
+void
+sbr_dec (HANDLE_SBR_DEC hSbrDec, /*!< handle to Decoder channel */
+ INT_PCM *timeIn, /*!< pointer to input time signal */
+ INT_PCM *timeOut, /*!< pointer to output time signal */
+ HANDLE_SBR_DEC hSbrDecRight, /*!< handle to Decoder channel right */
+ INT_PCM *timeOutRight, /*!< pointer to output time signal */
+ const int strideIn, /*!< Time data traversal strideIn */
+ const int strideOut, /*!< Time data traversal strideOut */
+ HANDLE_SBR_HEADER_DATA hHeaderData,/*!< Static control data */
+ HANDLE_SBR_FRAME_DATA hFrameData, /*!< Control data of current frame */
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData, /*!< Some control data of last frame */
+ const int applyProcessing, /*!< Flag for SBR operation */
+ HANDLE_PS_DEC h_ps_d,
+ const UINT flags
+ );
+
+
+
+SBR_ERROR
+createSbrDec (SBR_CHANNEL * hSbrChannel,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ TRANSPOSER_SETTINGS *pSettings,
+ const int downsampleFac,
+ const UINT qmfFlags,
+ const UINT flags,
+ const int overlap,
+ int chan);
+
+int
+deleteSbrDec (SBR_CHANNEL * hSbrChannel);
+
+SBR_ERROR
+resetSbrDec (HANDLE_SBR_DEC hSbrDec,
+ HANDLE_SBR_HEADER_DATA hHeaderData,
+ HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,
+ const int useLP,
+ const int downsampleFac);
+
+#endif
diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp
new file mode 100644
index 0000000..ee95e01
--- /dev/null
+++ b/libSBRdec/src/sbr_ram.cpp
@@ -0,0 +1,194 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Memory layout
+
+
+ This module declares all static and dynamic memory spaces
+*/
+
+#include "sbr_ram.h"
+
+
+
+
+#define WORKBUFFER1_TAG 0
+#define WORKBUFFER2_TAG 1
+
+/*!
+ \name StaticSbrData
+
+ Static memory areas, must not be overwritten in other sections of the decoder
+*/
+/* @{ */
+
+/*! SBR Decoder main structure */
+C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1)
+/*! SBR Decoder element data <br>
+ Dimension: (4) */
+C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (4))
+/*! SBR Decoder individual channel data <br>
+ Dimension: (6) */
+C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (6)+1)
+
+/*! Filter states for QMF-synthesis. <br>
+ Dimension: #(6) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
+C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (6)+1, SECT_DATA_L1)
+
+/*! Delayed spectral data needed for the dynamic framing of SBR.
+ For mp3PRO, 1/3 of a frame is buffered (#(6) 6) */
+C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (6)+1)
+
+/*! Static Data of PS */
+
+C_ALLOC_MEM(Ram_ps_dec, PS_DEC, 1)
+
+
+/* @} */
+
+
+/*!
+ \name DynamicSbrData
+
+ Dynamic memory areas, might be reused in other algorithm sections,
+ e.g. the core decoder
+ <br>
+ Depending on the mode set by DONT_USE_CORE_WORKBUFFER, workbuffers are
+ defined additionally to the CoreWorkbuffer.
+ <br>
+ The size of WorkBuffers is ((1024)/(32))*(64) = 2048.
+ <br>
+ WorkBuffer2 is a pointer to the CoreWorkBuffer wich is reused here in the SBR part. In case of
+ DONT_USE_CORE_WORKBUFFER, the CoreWorkbuffer is not used and the according
+ Workbuffer2 is defined locally in this file.
+ <br>
+ WorkBuffer1 is reused in the AAC core (-> aacdecoder.cpp, aac_ram.cpp)
+ <br>
+
+ Use of WorkBuffers:
+ <pre>
+
+ -------------------------------------------------------------
+ AAC core:
+
+ CoreWorkbuffer: spectral coefficients
+ WorkBuffer1: CAacDecoderChannelInfo, CAacDecoderDynamicData
+
+ -------------------------------------------------------------
+ SBR part:
+ ----------------------------------------------
+ Low Power Mode (useLP=1 or LOW_POWER_SBR_ONLY), see assignLcTimeSlots()
+
+ SLOT_BASED_PROTOTYPE_SYN_FILTER
+
+ WorkBuffer1 WorkBuffer2(=CoreWorkbuffer)
+ ________________ ________________
+ | RealLeft | | RealRight |
+ |________________| |________________|
+
+ ----------------------------------------------
+ High Quality Mode (!LOW_POWER_SBR_ONLY and useLP=0), see assignHqTimeSlots()
+
+ SLOTBASED_PS
+
+ WorkBuffer1 WorkBuffer2(=CoreWorkbuffer)
+ ________________ ________________
+ | Real/Imag | interleaved | Real/Imag | interleaved
+ |________________| first half actual ch |________________| second half actual ch
+
+ -------------------------------------------------------------
+
+ </pre>
+
+*/
+/* @{ */
+C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L1, WORKBUFFER1_TAG)
+C_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL, ((1024)/(32))*(64), SECT_DATA_L2, WORKBUFFER2_TAG)
+
+/* @} */
+
+
+
+
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
new file mode 100644
index 0000000..5469e51
--- /dev/null
+++ b/libSBRdec/src/sbr_ram.h
@@ -0,0 +1,158 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+\brief Memory layout
+
+*/
+#ifndef _SBR_RAM_H_
+#define _SBR_RAM_H_
+
+#include "sbrdecoder.h"
+
+#include "env_extr.h"
+#include "sbr_dec.h"
+
+
+
+#define SBRDEC_MAX_CH_PER_ELEMENT (2)
+
+typedef struct
+{
+ SBR_CHANNEL *pSbrChannel[SBRDEC_MAX_CH_PER_ELEMENT];
+ TRANSPOSER_SETTINGS transposerSettings; /* Common transport settings for each individual channel of an element */
+ HANDLE_FDK_BITSTREAM hBs;
+
+ MP4_ELEMENT_ID elementID; /* Element ID set during initialization. Can be used for concealment */
+ int nChannels; /* Number of elements output channels (=2 in case of PS) */
+
+ UCHAR frameErrorFlag[(1)+1]; /* Frame error status (for every slot in the delay line).
+ Will be copied into header at the very beginning of decodeElement() routine. */
+
+ UCHAR useFrameSlot; /* Index which defines which slot will be decoded/filled next (used with additional delay) */
+ UCHAR useHeaderSlot[(1)+1]; /* Index array that provides the link between header and frame data
+ (important when processing with additional delay). */
+} SBR_DECODER_ELEMENT;
+
+
+struct SBR_DECODER_INSTANCE
+{
+ SBR_DECODER_ELEMENT *pSbrElement[(4)];
+ SBR_HEADER_DATA sbrHeader[(4)][(1)+1]; /* Sbr header for each individual channel of an element */
+
+ FIXP_DBL *workBuffer1;
+ FIXP_DBL *workBuffer2;
+
+ HANDLE_PS_DEC hParametricStereoDec;
+
+ /* Global parameters */
+ AUDIO_OBJECT_TYPE coreCodec; /* AOT of core codec */
+ int numSbrElements;
+ int numSbrChannels;
+ INT sampleRateIn; /* SBR decoder input sampling rate; might be different than the transposer input sampling rate. */
+ INT sampleRateOut; /* Sampling rate of the SBR decoder output audio samples. */
+ USHORT codecFrameSize;
+ UCHAR synDownsampleFac;
+ UCHAR numDelayFrames; /* The current number of additional delay frames used for processing. */
+
+ UINT flags;
+
+};
+
+H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT)
+H_ALLOC_MEM(Ram_SbrDecChannel, SBR_CHANNEL)
+H_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE)
+
+H_ALLOC_MEM(Ram_sbr_QmfStatesSynthesis, FIXP_QSS)
+H_ALLOC_MEM(Ram_sbr_OverlapBuffer, FIXP_DBL)
+
+
+H_ALLOC_MEM(Ram_ps_dec, PS_DEC)
+
+
+H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer1, FIXP_DBL)
+H_ALLOC_MEM_OVERLAY(Ram_SbrDecWorkBuffer2, FIXP_DBL)
+
+
+#endif /* _SBR_RAM_H_ */
diff --git a/libSBRdec/src/sbr_rom.cpp b/libSBRdec/src/sbr_rom.cpp
new file mode 100644
index 0000000..8520b14
--- /dev/null
+++ b/libSBRdec/src/sbr_rom.cpp
@@ -0,0 +1,1412 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Definition of constant tables
+
+
+ This module contains most of the constant data that can be stored in ROM.
+*/
+
+#include "sbr_rom.h"
+
+
+
+
+/*!
+ \name StartStopBands
+ \brief Start and stop subbands of the highband.
+
+ k_o = startMin + offset[bs_start_freq];
+ startMin = {3000,4000,5000} * (128/FS_sbr) / FS_sbr < 32Khz, 32Khz <= FS_sbr < 64KHz, 64KHz <= FS_sbr
+ The stop subband can also be calculated to save memory by defining #CALC_STOP_BAND.
+*/
+//@{
+const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16] = {16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 26, 28, 30};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16] = {11, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16] = {10, 12, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 25, 27, 29, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16] = { 8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16] = { 7, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31};
+//@}
+
+
+/*!
+ \name Whitening
+ \brief Coefficients for spectral whitening in the transposer
+*/
+//@{
+/*! Assignment of whitening tuning depending on the crossover frequency */
+const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES] = {
+ 0,
+ 5000,
+ 6000,
+ 6500,
+ 7000,
+ 7500,
+ 8000,
+ 9000,
+ 10000
+};
+
+/*!
+ \brief Whithening levels tuning table
+
+ With the current tuning, there are some redundant entries:
+
+ \li NUM_WHFACTOR_TABLE_ENTRIES can be reduced by 3,
+ \li the first coloumn can be eliminated.
+
+*/
+const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6] = {
+ /* OFF_LEVEL, TRANSITION_LEVEL, LOW_LEVEL, MID_LEVEL, HIGH_LEVEL */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* < 5000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 5000 < 6000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6000 < 6500 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 6500 < 7000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7000 < 7500 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 7500 < 8000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 8000 < 9000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* 9000 < 10000 */
+ { FL2FXCONST_DBL(0.00f), FL2FXCONST_DBL(0.6f), FL2FXCONST_DBL(0.75f), FL2FXCONST_DBL(0.90f), FL2FXCONST_DBL(0.98f)}, /* > 10000 */
+};
+
+
+//@}
+
+
+/*!
+ \name EnvAdj
+ \brief Constants and tables used for envelope adjustment
+*/
+//@{
+
+/*! Mantissas of gain limits */
+const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4] =
+{
+ FL2FXCONST_SGL(0.5011932025f), /*!< -3 dB. Gain limit when limiterGains in frameData is 0 */
+ FL2FXCONST_SGL(0.5f), /*!< 0 dB. Gain limit when limiterGains in frameData is 1 */
+ FL2FXCONST_SGL(0.9976346258f), /*!< +3 dB. Gain limit when limiterGains in frameData is 2 */
+ FL2FXCONST_SGL(0.6776263578f) /*!< Inf. Gain limit when limiterGains in frameData is 3 */
+};
+
+/*! Exponents of gain limits */
+const UCHAR FDK_sbrDecoder_sbr_limGains_e[4] =
+{
+ 0, 1, 1, 67
+};
+
+/*! Constants for calculating the number of limiter bands */
+const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] =
+{
+ FL2FXCONST_SGL(1.0f / 4.0f),
+ FL2FXCONST_SGL(1.2f / 4.0f),
+ FL2FXCONST_SGL(2.0f / 4.0f),
+ FL2FXCONST_SGL(3.0f / 4.0f)
+};
+
+/*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */
+const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = {
+ FL2FXCONST_SGL(0.66666666666666f),
+ FL2FXCONST_SGL(0.36516383427084f),
+ FL2FXCONST_SGL(0.14699433520835f),
+ FL2FXCONST_SGL(0.03183050093751f)
+};
+
+
+/*! Real and imaginary part of random noise which will be modulated
+ to the desired level. An accuracy of 13 bits is sufficient for these
+ random numbers.
+*/
+const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2] = {
+ { FL2FXCONST_SGL(-0.99948153278296f / 8.0), FL2FXCONST_SGL(-0.59483417516607f / 8.0) },
+ { FL2FXCONST_SGL( 0.97113454393991f / 8.0), FL2FXCONST_SGL(-0.67528515225647f / 8.0) },
+ { FL2FXCONST_SGL( 0.14130051758487f / 8.0), FL2FXCONST_SGL(-0.95090983575689f / 8.0) },
+ { FL2FXCONST_SGL(-0.47005496701697f / 8.0), FL2FXCONST_SGL(-0.37340549728647f / 8.0) },
+ { FL2FXCONST_SGL( 0.80705063769351f / 8.0), FL2FXCONST_SGL( 0.29653668284408f / 8.0) },
+ { FL2FXCONST_SGL(-0.38981478896926f / 8.0), FL2FXCONST_SGL( 0.89572605717087f / 8.0) },
+ { FL2FXCONST_SGL(-0.01053049862020f / 8.0), FL2FXCONST_SGL(-0.66959058036166f / 8.0) },
+ { FL2FXCONST_SGL(-0.91266367957293f / 8.0), FL2FXCONST_SGL(-0.11522938140034f / 8.0) },
+ { FL2FXCONST_SGL( 0.54840422910309f / 8.0), FL2FXCONST_SGL( 0.75221367176302f / 8.0) },
+ { FL2FXCONST_SGL( 0.40009252867955f / 8.0), FL2FXCONST_SGL(-0.98929400334421f / 8.0) },
+ { FL2FXCONST_SGL(-0.99867974711855f / 8.0), FL2FXCONST_SGL(-0.88147068645358f / 8.0) },
+ { FL2FXCONST_SGL(-0.95531076805040f / 8.0), FL2FXCONST_SGL( 0.90908757154593f / 8.0) },
+ { FL2FXCONST_SGL(-0.45725933317144f / 8.0), FL2FXCONST_SGL(-0.56716323646760f / 8.0) },
+ { FL2FXCONST_SGL(-0.72929675029275f / 8.0), FL2FXCONST_SGL(-0.98008272727324f / 8.0) },
+ { FL2FXCONST_SGL( 0.75622801399036f / 8.0), FL2FXCONST_SGL( 0.20950329995549f / 8.0) },
+ { FL2FXCONST_SGL( 0.07069442601050f / 8.0), FL2FXCONST_SGL(-0.78247898470706f / 8.0) },
+ { FL2FXCONST_SGL( 0.74496252926055f / 8.0), FL2FXCONST_SGL(-0.91169004445807f / 8.0) },
+ { FL2FXCONST_SGL(-0.96440182703856f / 8.0), FL2FXCONST_SGL(-0.94739918296622f / 8.0) },
+ { FL2FXCONST_SGL( 0.30424629369539f / 8.0), FL2FXCONST_SGL(-0.49438267012479f / 8.0) },
+ { FL2FXCONST_SGL( 0.66565033746925f / 8.0), FL2FXCONST_SGL( 0.64652935542491f / 8.0) },
+ { FL2FXCONST_SGL( 0.91697008020594f / 8.0), FL2FXCONST_SGL( 0.17514097332009f / 8.0) },
+ { FL2FXCONST_SGL(-0.70774918760427f / 8.0), FL2FXCONST_SGL( 0.52548653416543f / 8.0) },
+ { FL2FXCONST_SGL(-0.70051415345560f / 8.0), FL2FXCONST_SGL(-0.45340028808763f / 8.0) },
+ { FL2FXCONST_SGL(-0.99496513054797f / 8.0), FL2FXCONST_SGL(-0.90071908066973f / 8.0) },
+ { FL2FXCONST_SGL( 0.98164490790123f / 8.0), FL2FXCONST_SGL(-0.77463155528697f / 8.0) },
+ { FL2FXCONST_SGL(-0.54671580548181f / 8.0), FL2FXCONST_SGL(-0.02570928536004f / 8.0) },
+ { FL2FXCONST_SGL(-0.01689629065389f / 8.0), FL2FXCONST_SGL( 0.00287506445732f / 8.0) },
+ { FL2FXCONST_SGL(-0.86110349531986f / 8.0), FL2FXCONST_SGL( 0.42548583726477f / 8.0) },
+ { FL2FXCONST_SGL(-0.98892980586032f / 8.0), FL2FXCONST_SGL(-0.87881132267556f / 8.0) },
+ { FL2FXCONST_SGL( 0.51756627678691f / 8.0), FL2FXCONST_SGL( 0.66926784710139f / 8.0) },
+ { FL2FXCONST_SGL(-0.99635026409640f / 8.0), FL2FXCONST_SGL(-0.58107730574765f / 8.0) },
+ { FL2FXCONST_SGL(-0.99969370862163f / 8.0), FL2FXCONST_SGL( 0.98369989360250f / 8.0) },
+ { FL2FXCONST_SGL( 0.55266258627194f / 8.0), FL2FXCONST_SGL( 0.59449057465591f / 8.0) },
+ { FL2FXCONST_SGL( 0.34581177741673f / 8.0), FL2FXCONST_SGL( 0.94879421061866f / 8.0) },
+ { FL2FXCONST_SGL( 0.62664209577999f / 8.0), FL2FXCONST_SGL(-0.74402970906471f / 8.0) },
+ { FL2FXCONST_SGL(-0.77149701404973f / 8.0), FL2FXCONST_SGL(-0.33883658042801f / 8.0) },
+ { FL2FXCONST_SGL(-0.91592244254432f / 8.0), FL2FXCONST_SGL( 0.03687901376713f / 8.0) },
+ { FL2FXCONST_SGL(-0.76285492357887f / 8.0), FL2FXCONST_SGL(-0.91371867919124f / 8.0) },
+ { FL2FXCONST_SGL( 0.79788337195331f / 8.0), FL2FXCONST_SGL(-0.93180971199849f / 8.0) },
+ { FL2FXCONST_SGL( 0.54473080610200f / 8.0), FL2FXCONST_SGL(-0.11919206037186f / 8.0) },
+ { FL2FXCONST_SGL(-0.85639281671058f / 8.0), FL2FXCONST_SGL( 0.42429854760451f / 8.0) },
+ { FL2FXCONST_SGL(-0.92882402971423f / 8.0), FL2FXCONST_SGL( 0.27871809078609f / 8.0) },
+ { FL2FXCONST_SGL(-0.11708371046774f / 8.0), FL2FXCONST_SGL(-0.99800843444966f / 8.0) },
+ { FL2FXCONST_SGL( 0.21356749817493f / 8.0), FL2FXCONST_SGL(-0.90716295627033f / 8.0) },
+ { FL2FXCONST_SGL(-0.76191692573909f / 8.0), FL2FXCONST_SGL( 0.99768118356265f / 8.0) },
+ { FL2FXCONST_SGL( 0.98111043100884f / 8.0), FL2FXCONST_SGL(-0.95854459734407f / 8.0) },
+ { FL2FXCONST_SGL(-0.85913269895572f / 8.0), FL2FXCONST_SGL( 0.95766566168880f / 8.0) },
+ { FL2FXCONST_SGL(-0.93307242253692f / 8.0), FL2FXCONST_SGL( 0.49431757696466f / 8.0) },
+ { FL2FXCONST_SGL( 0.30485754879632f / 8.0), FL2FXCONST_SGL(-0.70540034357529f / 8.0) },
+ { FL2FXCONST_SGL( 0.85289650925190f / 8.0), FL2FXCONST_SGL( 0.46766131791044f / 8.0) },
+ { FL2FXCONST_SGL( 0.91328082618125f / 8.0), FL2FXCONST_SGL(-0.99839597361769f / 8.0) },
+ { FL2FXCONST_SGL(-0.05890199924154f / 8.0), FL2FXCONST_SGL( 0.70741827819497f / 8.0) },
+ { FL2FXCONST_SGL( 0.28398686150148f / 8.0), FL2FXCONST_SGL( 0.34633555702188f / 8.0) },
+ { FL2FXCONST_SGL( 0.95258164539612f / 8.0), FL2FXCONST_SGL(-0.54893416026939f / 8.0) },
+ { FL2FXCONST_SGL(-0.78566324168507f / 8.0), FL2FXCONST_SGL(-0.75568541079691f / 8.0) },
+ { FL2FXCONST_SGL(-0.95789495447877f / 8.0), FL2FXCONST_SGL(-0.20423194696966f / 8.0) },
+ { FL2FXCONST_SGL( 0.82411158711197f / 8.0), FL2FXCONST_SGL( 0.96654618432562f / 8.0) },
+ { FL2FXCONST_SGL(-0.65185446735885f / 8.0), FL2FXCONST_SGL(-0.88734990773289f / 8.0) },
+ { FL2FXCONST_SGL(-0.93643603134666f / 8.0), FL2FXCONST_SGL( 0.99870790442385f / 8.0) },
+ { FL2FXCONST_SGL( 0.91427159529618f / 8.0), FL2FXCONST_SGL(-0.98290505544444f / 8.0) },
+ { FL2FXCONST_SGL(-0.70395684036886f / 8.0), FL2FXCONST_SGL( 0.58796798221039f / 8.0) },
+ { FL2FXCONST_SGL( 0.00563771969365f / 8.0), FL2FXCONST_SGL( 0.61768196727244f / 8.0) },
+ { FL2FXCONST_SGL( 0.89065051931895f / 8.0), FL2FXCONST_SGL( 0.52783352697585f / 8.0) },
+ { FL2FXCONST_SGL(-0.68683707712762f / 8.0), FL2FXCONST_SGL( 0.80806944710339f / 8.0) },
+ { FL2FXCONST_SGL( 0.72165342518718f / 8.0), FL2FXCONST_SGL(-0.69259857349564f / 8.0) },
+ { FL2FXCONST_SGL(-0.62928247730667f / 8.0), FL2FXCONST_SGL( 0.13627037407335f / 8.0) },
+ { FL2FXCONST_SGL( 0.29938434065514f / 8.0), FL2FXCONST_SGL(-0.46051329682246f / 8.0) },
+ { FL2FXCONST_SGL(-0.91781958879280f / 8.0), FL2FXCONST_SGL(-0.74012716684186f / 8.0) },
+ { FL2FXCONST_SGL( 0.99298717043688f / 8.0), FL2FXCONST_SGL( 0.40816610075661f / 8.0) },
+ { FL2FXCONST_SGL( 0.82368298622748f / 8.0), FL2FXCONST_SGL(-0.74036047190173f / 8.0) },
+ { FL2FXCONST_SGL(-0.98512833386833f / 8.0), FL2FXCONST_SGL(-0.99972330709594f / 8.0) },
+ { FL2FXCONST_SGL(-0.95915368242257f / 8.0), FL2FXCONST_SGL(-0.99237800466040f / 8.0) },
+ { FL2FXCONST_SGL(-0.21411126572790f / 8.0), FL2FXCONST_SGL(-0.93424819052545f / 8.0) },
+ { FL2FXCONST_SGL(-0.68821476106884f / 8.0), FL2FXCONST_SGL(-0.26892306315457f / 8.0) },
+ { FL2FXCONST_SGL( 0.91851997982317f / 8.0), FL2FXCONST_SGL( 0.09358228901785f / 8.0) },
+ { FL2FXCONST_SGL(-0.96062769559127f / 8.0), FL2FXCONST_SGL( 0.36099095133739f / 8.0) },
+ { FL2FXCONST_SGL( 0.51646184922287f / 8.0), FL2FXCONST_SGL(-0.71373332873917f / 8.0) },
+ { FL2FXCONST_SGL( 0.61130721139669f / 8.0), FL2FXCONST_SGL( 0.46950141175917f / 8.0) },
+ { FL2FXCONST_SGL( 0.47336129371299f / 8.0), FL2FXCONST_SGL(-0.27333178296162f / 8.0) },
+ { FL2FXCONST_SGL( 0.90998308703519f / 8.0), FL2FXCONST_SGL( 0.96715662938132f / 8.0) },
+ { FL2FXCONST_SGL( 0.44844799194357f / 8.0), FL2FXCONST_SGL( 0.99211574628306f / 8.0) },
+ { FL2FXCONST_SGL( 0.66614891079092f / 8.0), FL2FXCONST_SGL( 0.96590176169121f / 8.0) },
+ { FL2FXCONST_SGL( 0.74922239129237f / 8.0), FL2FXCONST_SGL(-0.89879858826087f / 8.0) },
+ { FL2FXCONST_SGL(-0.99571588506485f / 8.0), FL2FXCONST_SGL( 0.52785521494349f / 8.0) },
+ { FL2FXCONST_SGL( 0.97401082477563f / 8.0), FL2FXCONST_SGL(-0.16855870075190f / 8.0) },
+ { FL2FXCONST_SGL( 0.72683747733879f / 8.0), FL2FXCONST_SGL(-0.48060774432251f / 8.0) },
+ { FL2FXCONST_SGL( 0.95432193457128f / 8.0), FL2FXCONST_SGL( 0.68849603408441f / 8.0) },
+ { FL2FXCONST_SGL(-0.72962208425191f / 8.0), FL2FXCONST_SGL(-0.76608443420917f / 8.0) },
+ { FL2FXCONST_SGL(-0.85359479233537f / 8.0), FL2FXCONST_SGL( 0.88738125901579f / 8.0) },
+ { FL2FXCONST_SGL(-0.81412430338535f / 8.0), FL2FXCONST_SGL(-0.97480768049637f / 8.0) },
+ { FL2FXCONST_SGL(-0.87930772356786f / 8.0), FL2FXCONST_SGL( 0.74748307690436f / 8.0) },
+ { FL2FXCONST_SGL(-0.71573331064977f / 8.0), FL2FXCONST_SGL(-0.98570608178923f / 8.0) },
+ { FL2FXCONST_SGL( 0.83524300028228f / 8.0), FL2FXCONST_SGL( 0.83702537075163f / 8.0) },
+ { FL2FXCONST_SGL(-0.48086065601423f / 8.0), FL2FXCONST_SGL(-0.98848504923531f / 8.0) },
+ { FL2FXCONST_SGL( 0.97139128574778f / 8.0), FL2FXCONST_SGL( 0.80093621198236f / 8.0) },
+ { FL2FXCONST_SGL( 0.51992825347895f / 8.0), FL2FXCONST_SGL( 0.80247631400510f / 8.0) },
+ { FL2FXCONST_SGL(-0.00848591195325f / 8.0), FL2FXCONST_SGL(-0.76670128000486f / 8.0) },
+ { FL2FXCONST_SGL(-0.70294374303036f / 8.0), FL2FXCONST_SGL( 0.55359910445577f / 8.0) },
+ { FL2FXCONST_SGL(-0.95894428168140f / 8.0), FL2FXCONST_SGL(-0.43265504344783f / 8.0) },
+ { FL2FXCONST_SGL( 0.97079252950321f / 8.0), FL2FXCONST_SGL( 0.09325857238682f / 8.0) },
+ { FL2FXCONST_SGL(-0.92404293670797f / 8.0), FL2FXCONST_SGL( 0.85507704027855f / 8.0) },
+ { FL2FXCONST_SGL(-0.69506469500450f / 8.0), FL2FXCONST_SGL( 0.98633412625459f / 8.0) },
+ { FL2FXCONST_SGL( 0.26559203620024f / 8.0), FL2FXCONST_SGL( 0.73314307966524f / 8.0) },
+ { FL2FXCONST_SGL( 0.28038443336943f / 8.0), FL2FXCONST_SGL( 0.14537913654427f / 8.0) },
+ { FL2FXCONST_SGL(-0.74138124825523f / 8.0), FL2FXCONST_SGL( 0.99310339807762f / 8.0) },
+ { FL2FXCONST_SGL(-0.01752795995444f / 8.0), FL2FXCONST_SGL(-0.82616635284178f / 8.0) },
+ { FL2FXCONST_SGL(-0.55126773094930f / 8.0), FL2FXCONST_SGL(-0.98898543862153f / 8.0) },
+ { FL2FXCONST_SGL( 0.97960898850996f / 8.0), FL2FXCONST_SGL(-0.94021446752851f / 8.0) },
+ { FL2FXCONST_SGL(-0.99196309146936f / 8.0), FL2FXCONST_SGL( 0.67019017358456f / 8.0) },
+ { FL2FXCONST_SGL(-0.67684928085260f / 8.0), FL2FXCONST_SGL( 0.12631491649378f / 8.0) },
+ { FL2FXCONST_SGL( 0.09140039465500f / 8.0), FL2FXCONST_SGL(-0.20537731453108f / 8.0) },
+ { FL2FXCONST_SGL(-0.71658965751996f / 8.0), FL2FXCONST_SGL(-0.97788200391224f / 8.0) },
+ { FL2FXCONST_SGL( 0.81014640078925f / 8.0), FL2FXCONST_SGL( 0.53722648362443f / 8.0) },
+ { FL2FXCONST_SGL( 0.40616991671205f / 8.0), FL2FXCONST_SGL(-0.26469008598449f / 8.0) },
+ { FL2FXCONST_SGL(-0.67680188682972f / 8.0), FL2FXCONST_SGL( 0.94502052337695f / 8.0) },
+ { FL2FXCONST_SGL( 0.86849774348749f / 8.0), FL2FXCONST_SGL(-0.18333598647899f / 8.0) },
+ { FL2FXCONST_SGL(-0.99500381284851f / 8.0), FL2FXCONST_SGL(-0.02634122068550f / 8.0) },
+ { FL2FXCONST_SGL( 0.84329189340667f / 8.0), FL2FXCONST_SGL( 0.10406957462213f / 8.0) },
+ { FL2FXCONST_SGL(-0.09215968531446f / 8.0), FL2FXCONST_SGL( 0.69540012101253f / 8.0) },
+ { FL2FXCONST_SGL( 0.99956173327206f / 8.0), FL2FXCONST_SGL(-0.12358542001404f / 8.0) },
+ { FL2FXCONST_SGL(-0.79732779473535f / 8.0), FL2FXCONST_SGL(-0.91582524736159f / 8.0) },
+ { FL2FXCONST_SGL( 0.96349973642406f / 8.0), FL2FXCONST_SGL( 0.96640458041000f / 8.0) },
+ { FL2FXCONST_SGL(-0.79942778496547f / 8.0), FL2FXCONST_SGL( 0.64323902822857f / 8.0) },
+ { FL2FXCONST_SGL(-0.11566039853896f / 8.0), FL2FXCONST_SGL( 0.28587846253726f / 8.0) },
+ { FL2FXCONST_SGL(-0.39922954514662f / 8.0), FL2FXCONST_SGL( 0.94129601616966f / 8.0) },
+ { FL2FXCONST_SGL( 0.99089197565987f / 8.0), FL2FXCONST_SGL(-0.92062625581587f / 8.0) },
+ { FL2FXCONST_SGL( 0.28631285179909f / 8.0), FL2FXCONST_SGL(-0.91035047143603f / 8.0) },
+ { FL2FXCONST_SGL(-0.83302725605608f / 8.0), FL2FXCONST_SGL(-0.67330410892084f / 8.0) },
+ { FL2FXCONST_SGL( 0.95404443402072f / 8.0), FL2FXCONST_SGL( 0.49162765398743f / 8.0) },
+ { FL2FXCONST_SGL(-0.06449863579434f / 8.0), FL2FXCONST_SGL( 0.03250560813135f / 8.0) },
+ { FL2FXCONST_SGL(-0.99575054486311f / 8.0), FL2FXCONST_SGL( 0.42389784469507f / 8.0) },
+ { FL2FXCONST_SGL(-0.65501142790847f / 8.0), FL2FXCONST_SGL( 0.82546114655624f / 8.0) },
+ { FL2FXCONST_SGL(-0.81254441908887f / 8.0), FL2FXCONST_SGL(-0.51627234660629f / 8.0) },
+ { FL2FXCONST_SGL(-0.99646369485481f / 8.0), FL2FXCONST_SGL( 0.84490533520752f / 8.0) },
+ { FL2FXCONST_SGL( 0.00287840603348f / 8.0), FL2FXCONST_SGL( 0.64768261158166f / 8.0) },
+ { FL2FXCONST_SGL( 0.70176989408455f / 8.0), FL2FXCONST_SGL(-0.20453028573322f / 8.0) },
+ { FL2FXCONST_SGL( 0.96361882270190f / 8.0), FL2FXCONST_SGL( 0.40706967140989f / 8.0) },
+ { FL2FXCONST_SGL(-0.68883758192426f / 8.0), FL2FXCONST_SGL( 0.91338958840772f / 8.0) },
+ { FL2FXCONST_SGL(-0.34875585502238f / 8.0), FL2FXCONST_SGL( 0.71472290693300f / 8.0) },
+ { FL2FXCONST_SGL( 0.91980081243087f / 8.0), FL2FXCONST_SGL( 0.66507455644919f / 8.0) },
+ { FL2FXCONST_SGL(-0.99009048343881f / 8.0), FL2FXCONST_SGL( 0.85868021604848f / 8.0) },
+ { FL2FXCONST_SGL( 0.68865791458395f / 8.0), FL2FXCONST_SGL( 0.55660316809678f / 8.0) },
+ { FL2FXCONST_SGL(-0.99484402129368f / 8.0), FL2FXCONST_SGL(-0.20052559254934f / 8.0) },
+ { FL2FXCONST_SGL( 0.94214511408023f / 8.0), FL2FXCONST_SGL(-0.99696425367461f / 8.0) },
+ { FL2FXCONST_SGL(-0.67414626793544f / 8.0), FL2FXCONST_SGL( 0.49548221180078f / 8.0) },
+ { FL2FXCONST_SGL(-0.47339353684664f / 8.0), FL2FXCONST_SGL(-0.85904328834047f / 8.0) },
+ { FL2FXCONST_SGL( 0.14323651387360f / 8.0), FL2FXCONST_SGL(-0.94145598222488f / 8.0) },
+ { FL2FXCONST_SGL(-0.29268293575672f / 8.0), FL2FXCONST_SGL( 0.05759224927952f / 8.0) },
+ { FL2FXCONST_SGL( 0.43793861458754f / 8.0), FL2FXCONST_SGL(-0.78904969892724f / 8.0) },
+ { FL2FXCONST_SGL(-0.36345126374441f / 8.0), FL2FXCONST_SGL( 0.64874435357162f / 8.0) },
+ { FL2FXCONST_SGL(-0.08750604656825f / 8.0), FL2FXCONST_SGL( 0.97686944362527f / 8.0) },
+ { FL2FXCONST_SGL(-0.96495267812511f / 8.0), FL2FXCONST_SGL(-0.53960305946511f / 8.0) },
+ { FL2FXCONST_SGL( 0.55526940659947f / 8.0), FL2FXCONST_SGL( 0.78891523734774f / 8.0) },
+ { FL2FXCONST_SGL( 0.73538215752630f / 8.0), FL2FXCONST_SGL( 0.96452072373404f / 8.0) },
+ { FL2FXCONST_SGL(-0.30889773919437f / 8.0), FL2FXCONST_SGL(-0.80664389776860f / 8.0) },
+ { FL2FXCONST_SGL( 0.03574995626194f / 8.0), FL2FXCONST_SGL(-0.97325616900959f / 8.0) },
+ { FL2FXCONST_SGL( 0.98720684660488f / 8.0), FL2FXCONST_SGL( 0.48409133691962f / 8.0) },
+ { FL2FXCONST_SGL(-0.81689296271203f / 8.0), FL2FXCONST_SGL(-0.90827703628298f / 8.0) },
+ { FL2FXCONST_SGL( 0.67866860118215f / 8.0), FL2FXCONST_SGL( 0.81284503870856f / 8.0) },
+ { FL2FXCONST_SGL(-0.15808569732583f / 8.0), FL2FXCONST_SGL( 0.85279555024382f / 8.0) },
+ { FL2FXCONST_SGL( 0.80723395114371f / 8.0), FL2FXCONST_SGL(-0.24717418514605f / 8.0) },
+ { FL2FXCONST_SGL( 0.47788757329038f / 8.0), FL2FXCONST_SGL(-0.46333147839295f / 8.0) },
+ { FL2FXCONST_SGL( 0.96367554763201f / 8.0), FL2FXCONST_SGL( 0.38486749303242f / 8.0) },
+ { FL2FXCONST_SGL(-0.99143875716818f / 8.0), FL2FXCONST_SGL(-0.24945277239809f / 8.0) },
+ { FL2FXCONST_SGL( 0.83081876925833f / 8.0), FL2FXCONST_SGL(-0.94780851414763f / 8.0) },
+ { FL2FXCONST_SGL(-0.58753191905341f / 8.0), FL2FXCONST_SGL( 0.01290772389163f / 8.0) },
+ { FL2FXCONST_SGL( 0.95538108220960f / 8.0), FL2FXCONST_SGL(-0.85557052096538f / 8.0) },
+ { FL2FXCONST_SGL(-0.96490920476211f / 8.0), FL2FXCONST_SGL(-0.64020970923102f / 8.0) },
+ { FL2FXCONST_SGL(-0.97327101028521f / 8.0), FL2FXCONST_SGL( 0.12378128133110f / 8.0) },
+ { FL2FXCONST_SGL( 0.91400366022124f / 8.0), FL2FXCONST_SGL( 0.57972471346930f / 8.0) },
+ { FL2FXCONST_SGL(-0.99925837363824f / 8.0), FL2FXCONST_SGL( 0.71084847864067f / 8.0) },
+ { FL2FXCONST_SGL(-0.86875903507313f / 8.0), FL2FXCONST_SGL(-0.20291699203564f / 8.0) },
+ { FL2FXCONST_SGL(-0.26240034795124f / 8.0), FL2FXCONST_SGL(-0.68264554369108f / 8.0) },
+ { FL2FXCONST_SGL(-0.24664412953388f / 8.0), FL2FXCONST_SGL(-0.87642273115183f / 8.0) },
+ { FL2FXCONST_SGL( 0.02416275806869f / 8.0), FL2FXCONST_SGL( 0.27192914288905f / 8.0) },
+ { FL2FXCONST_SGL( 0.82068619590515f / 8.0), FL2FXCONST_SGL(-0.85087787994476f / 8.0) },
+ { FL2FXCONST_SGL( 0.88547373760759f / 8.0), FL2FXCONST_SGL(-0.89636802901469f / 8.0) },
+ { FL2FXCONST_SGL(-0.18173078152226f / 8.0), FL2FXCONST_SGL(-0.26152145156800f / 8.0) },
+ { FL2FXCONST_SGL( 0.09355476558534f / 8.0), FL2FXCONST_SGL( 0.54845123045604f / 8.0) },
+ { FL2FXCONST_SGL(-0.54668414224090f / 8.0), FL2FXCONST_SGL( 0.95980774020221f / 8.0) },
+ { FL2FXCONST_SGL( 0.37050990604091f / 8.0), FL2FXCONST_SGL(-0.59910140383171f / 8.0) },
+ { FL2FXCONST_SGL(-0.70373594262891f / 8.0), FL2FXCONST_SGL( 0.91227665827081f / 8.0) },
+ { FL2FXCONST_SGL(-0.34600785879594f / 8.0), FL2FXCONST_SGL(-0.99441426144200f / 8.0) },
+ { FL2FXCONST_SGL(-0.68774481731008f / 8.0), FL2FXCONST_SGL(-0.30238837956299f / 8.0) },
+ { FL2FXCONST_SGL(-0.26843291251234f / 8.0), FL2FXCONST_SGL( 0.83115668004362f / 8.0) },
+ { FL2FXCONST_SGL( 0.49072334613242f / 8.0), FL2FXCONST_SGL(-0.45359708737775f / 8.0) },
+ { FL2FXCONST_SGL( 0.38975993093975f / 8.0), FL2FXCONST_SGL( 0.95515358099121f / 8.0) },
+ { FL2FXCONST_SGL(-0.97757125224150f / 8.0), FL2FXCONST_SGL( 0.05305894580606f / 8.0) },
+ { FL2FXCONST_SGL(-0.17325552859616f / 8.0), FL2FXCONST_SGL(-0.92770672250494f / 8.0) },
+ { FL2FXCONST_SGL( 0.99948035025744f / 8.0), FL2FXCONST_SGL( 0.58285545563426f / 8.0) },
+ { FL2FXCONST_SGL(-0.64946246527458f / 8.0), FL2FXCONST_SGL( 0.68645507104960f / 8.0) },
+ { FL2FXCONST_SGL(-0.12016920576437f / 8.0), FL2FXCONST_SGL(-0.57147322153312f / 8.0) },
+ { FL2FXCONST_SGL(-0.58947456517751f / 8.0), FL2FXCONST_SGL(-0.34847132454388f / 8.0) },
+ { FL2FXCONST_SGL(-0.41815140454465f / 8.0), FL2FXCONST_SGL( 0.16276422358861f / 8.0) },
+ { FL2FXCONST_SGL( 0.99885650204884f / 8.0), FL2FXCONST_SGL( 0.11136095490444f / 8.0) },
+ { FL2FXCONST_SGL(-0.56649614128386f / 8.0), FL2FXCONST_SGL(-0.90494866361587f / 8.0) },
+ { FL2FXCONST_SGL( 0.94138021032330f / 8.0), FL2FXCONST_SGL( 0.35281916733018f / 8.0) },
+ { FL2FXCONST_SGL(-0.75725076534641f / 8.0), FL2FXCONST_SGL( 0.53650549640587f / 8.0) },
+ { FL2FXCONST_SGL( 0.20541973692630f / 8.0), FL2FXCONST_SGL(-0.94435144369918f / 8.0) },
+ { FL2FXCONST_SGL( 0.99980371023351f / 8.0), FL2FXCONST_SGL( 0.79835913565599f / 8.0) },
+ { FL2FXCONST_SGL( 0.29078277605775f / 8.0), FL2FXCONST_SGL( 0.35393777921520f / 8.0) },
+ { FL2FXCONST_SGL(-0.62858772103030f / 8.0), FL2FXCONST_SGL( 0.38765693387102f / 8.0) },
+ { FL2FXCONST_SGL( 0.43440904467688f / 8.0), FL2FXCONST_SGL(-0.98546330463232f / 8.0) },
+ { FL2FXCONST_SGL(-0.98298583762390f / 8.0), FL2FXCONST_SGL( 0.21021524625209f / 8.0) },
+ { FL2FXCONST_SGL( 0.19513029146934f / 8.0), FL2FXCONST_SGL(-0.94239832251867f / 8.0) },
+ { FL2FXCONST_SGL(-0.95476662400101f / 8.0), FL2FXCONST_SGL( 0.98364554179143f / 8.0) },
+ { FL2FXCONST_SGL( 0.93379635304810f / 8.0), FL2FXCONST_SGL(-0.70881994583682f / 8.0) },
+ { FL2FXCONST_SGL(-0.85235410573336f / 8.0), FL2FXCONST_SGL(-0.08342347966410f / 8.0) },
+ { FL2FXCONST_SGL(-0.86425093011245f / 8.0), FL2FXCONST_SGL(-0.45795025029466f / 8.0) },
+ { FL2FXCONST_SGL( 0.38879779059045f / 8.0), FL2FXCONST_SGL( 0.97274429344593f / 8.0) },
+ { FL2FXCONST_SGL( 0.92045124735495f / 8.0), FL2FXCONST_SGL(-0.62433652524220f / 8.0) },
+ { FL2FXCONST_SGL( 0.89162532251878f / 8.0), FL2FXCONST_SGL( 0.54950955570563f / 8.0) },
+ { FL2FXCONST_SGL(-0.36834336949252f / 8.0), FL2FXCONST_SGL( 0.96458298020975f / 8.0) },
+ { FL2FXCONST_SGL( 0.93891760988045f / 8.0), FL2FXCONST_SGL(-0.89968353740388f / 8.0) },
+ { FL2FXCONST_SGL( 0.99267657565094f / 8.0), FL2FXCONST_SGL(-0.03757034316958f / 8.0) },
+ { FL2FXCONST_SGL(-0.94063471614176f / 8.0), FL2FXCONST_SGL( 0.41332338538963f / 8.0) },
+ { FL2FXCONST_SGL( 0.99740224117019f / 8.0), FL2FXCONST_SGL(-0.16830494996370f / 8.0) },
+ { FL2FXCONST_SGL(-0.35899413170555f / 8.0), FL2FXCONST_SGL(-0.46633226649613f / 8.0) },
+ { FL2FXCONST_SGL( 0.05237237274947f / 8.0), FL2FXCONST_SGL(-0.25640361602661f / 8.0) },
+ { FL2FXCONST_SGL( 0.36703583957424f / 8.0), FL2FXCONST_SGL(-0.38653265641875f / 8.0) },
+ { FL2FXCONST_SGL( 0.91653180367913f / 8.0), FL2FXCONST_SGL(-0.30587628726597f / 8.0) },
+ { FL2FXCONST_SGL( 0.69000803499316f / 8.0), FL2FXCONST_SGL( 0.90952171386132f / 8.0) },
+ { FL2FXCONST_SGL(-0.38658751133527f / 8.0), FL2FXCONST_SGL( 0.99501571208985f / 8.0) },
+ { FL2FXCONST_SGL(-0.29250814029851f / 8.0), FL2FXCONST_SGL( 0.37444994344615f / 8.0) },
+ { FL2FXCONST_SGL(-0.60182204677608f / 8.0), FL2FXCONST_SGL( 0.86779651036123f / 8.0) },
+ { FL2FXCONST_SGL(-0.97418588163217f / 8.0), FL2FXCONST_SGL( 0.96468523666475f / 8.0) },
+ { FL2FXCONST_SGL( 0.88461574003963f / 8.0), FL2FXCONST_SGL( 0.57508405276414f / 8.0) },
+ { FL2FXCONST_SGL( 0.05198933055162f / 8.0), FL2FXCONST_SGL( 0.21269661669964f / 8.0) },
+ { FL2FXCONST_SGL(-0.53499621979720f / 8.0), FL2FXCONST_SGL( 0.97241553731237f / 8.0) },
+ { FL2FXCONST_SGL(-0.49429560226497f / 8.0), FL2FXCONST_SGL( 0.98183865291903f / 8.0) },
+ { FL2FXCONST_SGL(-0.98935142339139f / 8.0), FL2FXCONST_SGL(-0.40249159006933f / 8.0) },
+ { FL2FXCONST_SGL(-0.98081380091130f / 8.0), FL2FXCONST_SGL(-0.72856895534041f / 8.0) },
+ { FL2FXCONST_SGL(-0.27338148835532f / 8.0), FL2FXCONST_SGL( 0.99950922447209f / 8.0) },
+ { FL2FXCONST_SGL( 0.06310802338302f / 8.0), FL2FXCONST_SGL(-0.54539587529618f / 8.0) },
+ { FL2FXCONST_SGL(-0.20461677199539f / 8.0), FL2FXCONST_SGL(-0.14209977628489f / 8.0) },
+ { FL2FXCONST_SGL( 0.66223843141647f / 8.0), FL2FXCONST_SGL( 0.72528579940326f / 8.0) },
+ { FL2FXCONST_SGL(-0.84764345483665f / 8.0), FL2FXCONST_SGL( 0.02372316801261f / 8.0) },
+ { FL2FXCONST_SGL(-0.89039863483811f / 8.0), FL2FXCONST_SGL( 0.88866581484602f / 8.0) },
+ { FL2FXCONST_SGL( 0.95903308477986f / 8.0), FL2FXCONST_SGL( 0.76744927173873f / 8.0) },
+ { FL2FXCONST_SGL( 0.73504123909879f / 8.0), FL2FXCONST_SGL(-0.03747203173192f / 8.0) },
+ { FL2FXCONST_SGL(-0.31744434966056f / 8.0), FL2FXCONST_SGL(-0.36834111883652f / 8.0) },
+ { FL2FXCONST_SGL(-0.34110827591623f / 8.0), FL2FXCONST_SGL( 0.40211222807691f / 8.0) },
+ { FL2FXCONST_SGL( 0.47803883714199f / 8.0), FL2FXCONST_SGL(-0.39423219786288f / 8.0) },
+ { FL2FXCONST_SGL( 0.98299195879514f / 8.0), FL2FXCONST_SGL( 0.01989791390047f / 8.0) },
+ { FL2FXCONST_SGL(-0.30963073129751f / 8.0), FL2FXCONST_SGL(-0.18076720599336f / 8.0) },
+ { FL2FXCONST_SGL( 0.99992588229018f / 8.0), FL2FXCONST_SGL(-0.26281872094289f / 8.0) },
+ { FL2FXCONST_SGL(-0.93149731080767f / 8.0), FL2FXCONST_SGL(-0.98313162570490f / 8.0) },
+ { FL2FXCONST_SGL( 0.99923472302773f / 8.0), FL2FXCONST_SGL(-0.80142993767554f / 8.0) },
+ { FL2FXCONST_SGL(-0.26024169633417f / 8.0), FL2FXCONST_SGL(-0.75999759855752f / 8.0) },
+ { FL2FXCONST_SGL(-0.35712514743563f / 8.0), FL2FXCONST_SGL( 0.19298963768574f / 8.0) },
+ { FL2FXCONST_SGL(-0.99899084509530f / 8.0), FL2FXCONST_SGL( 0.74645156992493f / 8.0) },
+ { FL2FXCONST_SGL( 0.86557171579452f / 8.0), FL2FXCONST_SGL( 0.55593866696299f / 8.0) },
+ { FL2FXCONST_SGL( 0.33408042438752f / 8.0), FL2FXCONST_SGL( 0.86185953874709f / 8.0) },
+ { FL2FXCONST_SGL( 0.99010736374716f / 8.0), FL2FXCONST_SGL( 0.04602397576623f / 8.0) },
+ { FL2FXCONST_SGL(-0.66694269691195f / 8.0), FL2FXCONST_SGL(-0.91643611810148f / 8.0) },
+ { FL2FXCONST_SGL( 0.64016792079480f / 8.0), FL2FXCONST_SGL( 0.15649530836856f / 8.0) },
+ { FL2FXCONST_SGL( 0.99570534804836f / 8.0), FL2FXCONST_SGL( 0.45844586038111f / 8.0) },
+ { FL2FXCONST_SGL(-0.63431466947340f / 8.0), FL2FXCONST_SGL( 0.21079116459234f / 8.0) },
+ { FL2FXCONST_SGL(-0.07706847005931f / 8.0), FL2FXCONST_SGL(-0.89581437101329f / 8.0) },
+ { FL2FXCONST_SGL( 0.98590090577724f / 8.0), FL2FXCONST_SGL( 0.88241721133981f / 8.0) },
+ { FL2FXCONST_SGL( 0.80099335254678f / 8.0), FL2FXCONST_SGL(-0.36851896710853f / 8.0) },
+ { FL2FXCONST_SGL( 0.78368131392666f / 8.0), FL2FXCONST_SGL( 0.45506999802597f / 8.0) },
+ { FL2FXCONST_SGL( 0.08707806671691f / 8.0), FL2FXCONST_SGL( 0.80938994918745f / 8.0) },
+ { FL2FXCONST_SGL(-0.86811883080712f / 8.0), FL2FXCONST_SGL( 0.39347308654705f / 8.0) },
+ { FL2FXCONST_SGL(-0.39466529740375f / 8.0), FL2FXCONST_SGL(-0.66809432114456f / 8.0) },
+ { FL2FXCONST_SGL( 0.97875325649683f / 8.0), FL2FXCONST_SGL(-0.72467840967746f / 8.0) },
+ { FL2FXCONST_SGL(-0.95038560288864f / 8.0), FL2FXCONST_SGL( 0.89563219587625f / 8.0) },
+ { FL2FXCONST_SGL( 0.17005239424212f / 8.0), FL2FXCONST_SGL( 0.54683053962658f / 8.0) },
+ { FL2FXCONST_SGL(-0.76910792026848f / 8.0), FL2FXCONST_SGL(-0.96226617549298f / 8.0) },
+ { FL2FXCONST_SGL( 0.99743281016846f / 8.0), FL2FXCONST_SGL( 0.42697157037567f / 8.0) },
+ { FL2FXCONST_SGL( 0.95437383549973f / 8.0), FL2FXCONST_SGL( 0.97002324109952f / 8.0) },
+ { FL2FXCONST_SGL( 0.99578905365569f / 8.0), FL2FXCONST_SGL(-0.54106826257356f / 8.0) },
+ { FL2FXCONST_SGL( 0.28058259829990f / 8.0), FL2FXCONST_SGL(-0.85361420634036f / 8.0) },
+ { FL2FXCONST_SGL( 0.85256524470573f / 8.0), FL2FXCONST_SGL(-0.64567607735589f / 8.0) },
+ { FL2FXCONST_SGL(-0.50608540105128f / 8.0), FL2FXCONST_SGL(-0.65846015480300f / 8.0) },
+ { FL2FXCONST_SGL(-0.97210735183243f / 8.0), FL2FXCONST_SGL(-0.23095213067791f / 8.0) },
+ { FL2FXCONST_SGL( 0.95424048234441f / 8.0), FL2FXCONST_SGL(-0.99240147091219f / 8.0) },
+ { FL2FXCONST_SGL(-0.96926570524023f / 8.0), FL2FXCONST_SGL( 0.73775654896574f / 8.0) },
+ { FL2FXCONST_SGL( 0.30872163214726f / 8.0), FL2FXCONST_SGL( 0.41514960556126f / 8.0) },
+ { FL2FXCONST_SGL(-0.24523839572639f / 8.0), FL2FXCONST_SGL( 0.63206633394807f / 8.0) },
+ { FL2FXCONST_SGL(-0.33813265086024f / 8.0), FL2FXCONST_SGL(-0.38661779441897f / 8.0) },
+ { FL2FXCONST_SGL(-0.05826828420146f / 8.0), FL2FXCONST_SGL(-0.06940774188029f / 8.0) },
+ { FL2FXCONST_SGL(-0.22898461455054f / 8.0), FL2FXCONST_SGL( 0.97054853316316f / 8.0) },
+ { FL2FXCONST_SGL(-0.18509915019881f / 8.0), FL2FXCONST_SGL( 0.47565762892084f / 8.0) },
+ { FL2FXCONST_SGL(-0.10488238045009f / 8.0), FL2FXCONST_SGL(-0.87769947402394f / 8.0) },
+ { FL2FXCONST_SGL(-0.71886586182037f / 8.0), FL2FXCONST_SGL( 0.78030982480538f / 8.0) },
+ { FL2FXCONST_SGL( 0.99793873738654f / 8.0), FL2FXCONST_SGL( 0.90041310491497f / 8.0) },
+ { FL2FXCONST_SGL( 0.57563307626120f / 8.0), FL2FXCONST_SGL(-0.91034337352097f / 8.0) },
+ { FL2FXCONST_SGL( 0.28909646383717f / 8.0), FL2FXCONST_SGL( 0.96307783970534f / 8.0) },
+ { FL2FXCONST_SGL( 0.42188998312520f / 8.0), FL2FXCONST_SGL( 0.48148651230437f / 8.0) },
+ { FL2FXCONST_SGL( 0.93335049681047f / 8.0), FL2FXCONST_SGL(-0.43537023883588f / 8.0) },
+ { FL2FXCONST_SGL(-0.97087374418267f / 8.0), FL2FXCONST_SGL( 0.86636445711364f / 8.0) },
+ { FL2FXCONST_SGL( 0.36722871286923f / 8.0), FL2FXCONST_SGL( 0.65291654172961f / 8.0) },
+ { FL2FXCONST_SGL(-0.81093025665696f / 8.0), FL2FXCONST_SGL( 0.08778370229363f / 8.0) },
+ { FL2FXCONST_SGL(-0.26240603062237f / 8.0), FL2FXCONST_SGL(-0.92774095379098f / 8.0) },
+ { FL2FXCONST_SGL( 0.83996497984604f / 8.0), FL2FXCONST_SGL( 0.55839849139647f / 8.0) },
+ { FL2FXCONST_SGL(-0.99909615720225f / 8.0), FL2FXCONST_SGL(-0.96024605713970f / 8.0) },
+ { FL2FXCONST_SGL( 0.74649464155061f / 8.0), FL2FXCONST_SGL( 0.12144893606462f / 8.0) },
+ { FL2FXCONST_SGL(-0.74774595569805f / 8.0), FL2FXCONST_SGL(-0.26898062008959f / 8.0) },
+ { FL2FXCONST_SGL( 0.95781667469567f / 8.0), FL2FXCONST_SGL(-0.79047927052628f / 8.0) },
+ { FL2FXCONST_SGL( 0.95472308713099f / 8.0), FL2FXCONST_SGL(-0.08588776019550f / 8.0) },
+ { FL2FXCONST_SGL( 0.48708332746299f / 8.0), FL2FXCONST_SGL( 0.99999041579432f / 8.0) },
+ { FL2FXCONST_SGL( 0.46332038247497f / 8.0), FL2FXCONST_SGL( 0.10964126185063f / 8.0) },
+ { FL2FXCONST_SGL(-0.76497004940162f / 8.0), FL2FXCONST_SGL( 0.89210929242238f / 8.0) },
+ { FL2FXCONST_SGL( 0.57397389364339f / 8.0), FL2FXCONST_SGL( 0.35289703373760f / 8.0) },
+ { FL2FXCONST_SGL( 0.75374316974495f / 8.0), FL2FXCONST_SGL( 0.96705214651335f / 8.0) },
+ { FL2FXCONST_SGL(-0.59174397685714f / 8.0), FL2FXCONST_SGL(-0.89405370422752f / 8.0) },
+ { FL2FXCONST_SGL( 0.75087906691890f / 8.0), FL2FXCONST_SGL(-0.29612672982396f / 8.0) },
+ { FL2FXCONST_SGL(-0.98607857336230f / 8.0), FL2FXCONST_SGL( 0.25034911730023f / 8.0) },
+ { FL2FXCONST_SGL(-0.40761056640505f / 8.0), FL2FXCONST_SGL(-0.90045573444695f / 8.0) },
+ { FL2FXCONST_SGL( 0.66929266740477f / 8.0), FL2FXCONST_SGL( 0.98629493401748f / 8.0) },
+ { FL2FXCONST_SGL(-0.97463695257310f / 8.0), FL2FXCONST_SGL(-0.00190223301301f / 8.0) },
+ { FL2FXCONST_SGL( 0.90145509409859f / 8.0), FL2FXCONST_SGL( 0.99781390365446f / 8.0) },
+ { FL2FXCONST_SGL(-0.87259289048043f / 8.0), FL2FXCONST_SGL( 0.99233587353666f / 8.0) },
+ { FL2FXCONST_SGL(-0.91529461447692f / 8.0), FL2FXCONST_SGL(-0.15698707534206f / 8.0) },
+ { FL2FXCONST_SGL(-0.03305738840705f / 8.0), FL2FXCONST_SGL(-0.37205262859764f / 8.0) },
+ { FL2FXCONST_SGL( 0.07223051368337f / 8.0), FL2FXCONST_SGL(-0.88805001733626f / 8.0) },
+ { FL2FXCONST_SGL( 0.99498012188353f / 8.0), FL2FXCONST_SGL( 0.97094358113387f / 8.0) },
+ { FL2FXCONST_SGL(-0.74904939500519f / 8.0), FL2FXCONST_SGL( 0.99985483641521f / 8.0) },
+ { FL2FXCONST_SGL( 0.04585228574211f / 8.0), FL2FXCONST_SGL( 0.99812337444082f / 8.0) },
+ { FL2FXCONST_SGL(-0.89054954257993f / 8.0), FL2FXCONST_SGL(-0.31791913188064f / 8.0) },
+ { FL2FXCONST_SGL(-0.83782144651251f / 8.0), FL2FXCONST_SGL( 0.97637632547466f / 8.0) },
+ { FL2FXCONST_SGL( 0.33454804933804f / 8.0), FL2FXCONST_SGL(-0.86231516800408f / 8.0) },
+ { FL2FXCONST_SGL(-0.99707579362824f / 8.0), FL2FXCONST_SGL( 0.93237990079441f / 8.0) },
+ { FL2FXCONST_SGL(-0.22827527843994f / 8.0), FL2FXCONST_SGL( 0.18874759397997f / 8.0) },
+ { FL2FXCONST_SGL( 0.67248046289143f / 8.0), FL2FXCONST_SGL(-0.03646211390569f / 8.0) },
+ { FL2FXCONST_SGL(-0.05146538187944f / 8.0), FL2FXCONST_SGL(-0.92599700120679f / 8.0) },
+ { FL2FXCONST_SGL( 0.99947295749905f / 8.0), FL2FXCONST_SGL( 0.93625229707912f / 8.0) },
+ { FL2FXCONST_SGL( 0.66951124390363f / 8.0), FL2FXCONST_SGL( 0.98905825623893f / 8.0) },
+ { FL2FXCONST_SGL(-0.99602956559179f / 8.0), FL2FXCONST_SGL(-0.44654715757688f / 8.0) },
+ { FL2FXCONST_SGL( 0.82104905483590f / 8.0), FL2FXCONST_SGL( 0.99540741724928f / 8.0) },
+ { FL2FXCONST_SGL( 0.99186510988782f / 8.0), FL2FXCONST_SGL( 0.72023001312947f / 8.0) },
+ { FL2FXCONST_SGL(-0.65284592392918f / 8.0), FL2FXCONST_SGL( 0.52186723253637f / 8.0) },
+ { FL2FXCONST_SGL( 0.93885443798188f / 8.0), FL2FXCONST_SGL(-0.74895312615259f / 8.0) },
+ { FL2FXCONST_SGL( 0.96735248738388f / 8.0), FL2FXCONST_SGL( 0.90891816978629f / 8.0) },
+ { FL2FXCONST_SGL(-0.22225968841114f / 8.0), FL2FXCONST_SGL( 0.57124029781228f / 8.0) },
+ { FL2FXCONST_SGL(-0.44132783753414f / 8.0), FL2FXCONST_SGL(-0.92688840659280f / 8.0) },
+ { FL2FXCONST_SGL(-0.85694974219574f / 8.0), FL2FXCONST_SGL( 0.88844532719844f / 8.0) },
+ { FL2FXCONST_SGL( 0.91783042091762f / 8.0), FL2FXCONST_SGL(-0.46356892383970f / 8.0) },
+ { FL2FXCONST_SGL( 0.72556974415690f / 8.0), FL2FXCONST_SGL(-0.99899555770747f / 8.0) },
+ { FL2FXCONST_SGL(-0.99711581834508f / 8.0), FL2FXCONST_SGL( 0.58211560180426f / 8.0) },
+ { FL2FXCONST_SGL( 0.77638976371966f / 8.0), FL2FXCONST_SGL( 0.94321834873819f / 8.0) },
+ { FL2FXCONST_SGL( 0.07717324253925f / 8.0), FL2FXCONST_SGL( 0.58638399856595f / 8.0) },
+ { FL2FXCONST_SGL(-0.56049829194163f / 8.0), FL2FXCONST_SGL( 0.82522301569036f / 8.0) },
+ { FL2FXCONST_SGL( 0.98398893639988f / 8.0), FL2FXCONST_SGL( 0.39467440420569f / 8.0) },
+ { FL2FXCONST_SGL( 0.47546946844938f / 8.0), FL2FXCONST_SGL( 0.68613044836811f / 8.0) },
+ { FL2FXCONST_SGL( 0.65675089314631f / 8.0), FL2FXCONST_SGL( 0.18331637134880f / 8.0) },
+ { FL2FXCONST_SGL( 0.03273375457980f / 8.0), FL2FXCONST_SGL(-0.74933109564108f / 8.0) },
+ { FL2FXCONST_SGL(-0.38684144784738f / 8.0), FL2FXCONST_SGL( 0.51337349030406f / 8.0) },
+ { FL2FXCONST_SGL(-0.97346267944545f / 8.0), FL2FXCONST_SGL(-0.96549364384098f / 8.0) },
+ { FL2FXCONST_SGL(-0.53282156061942f / 8.0), FL2FXCONST_SGL(-0.91423265091354f / 8.0) },
+ { FL2FXCONST_SGL( 0.99817310731176f / 8.0), FL2FXCONST_SGL( 0.61133572482148f / 8.0) },
+ { FL2FXCONST_SGL(-0.50254500772635f / 8.0), FL2FXCONST_SGL(-0.88829338134294f / 8.0) },
+ { FL2FXCONST_SGL( 0.01995873238855f / 8.0), FL2FXCONST_SGL( 0.85223515096765f / 8.0) },
+ { FL2FXCONST_SGL( 0.99930381973804f / 8.0), FL2FXCONST_SGL( 0.94578896296649f / 8.0) },
+ { FL2FXCONST_SGL( 0.82907767600783f / 8.0), FL2FXCONST_SGL(-0.06323442598128f / 8.0) },
+ { FL2FXCONST_SGL(-0.58660709669728f / 8.0), FL2FXCONST_SGL( 0.96840773806582f / 8.0) },
+ { FL2FXCONST_SGL(-0.17573736667267f / 8.0), FL2FXCONST_SGL(-0.48166920859485f / 8.0) },
+ { FL2FXCONST_SGL( 0.83434292401346f / 8.0), FL2FXCONST_SGL(-0.13023450646997f / 8.0) },
+ { FL2FXCONST_SGL( 0.05946491307025f / 8.0), FL2FXCONST_SGL( 0.20511047074866f / 8.0) },
+ { FL2FXCONST_SGL( 0.81505484574602f / 8.0), FL2FXCONST_SGL(-0.94685947861369f / 8.0) },
+ { FL2FXCONST_SGL(-0.44976380954860f / 8.0), FL2FXCONST_SGL( 0.40894572671545f / 8.0) },
+ { FL2FXCONST_SGL(-0.89746474625671f / 8.0), FL2FXCONST_SGL( 0.99846578838537f / 8.0) },
+ { FL2FXCONST_SGL( 0.39677256130792f / 8.0), FL2FXCONST_SGL(-0.74854668609359f / 8.0) },
+ { FL2FXCONST_SGL(-0.07588948563079f / 8.0), FL2FXCONST_SGL( 0.74096214084170f / 8.0) },
+ { FL2FXCONST_SGL( 0.76343198951445f / 8.0), FL2FXCONST_SGL( 0.41746629422634f / 8.0) },
+ { FL2FXCONST_SGL(-0.74490104699626f / 8.0), FL2FXCONST_SGL( 0.94725911744610f / 8.0) },
+ { FL2FXCONST_SGL( 0.64880119792759f / 8.0), FL2FXCONST_SGL( 0.41336660830571f / 8.0) },
+ { FL2FXCONST_SGL( 0.62319537462542f / 8.0), FL2FXCONST_SGL(-0.93098313552599f / 8.0) },
+ { FL2FXCONST_SGL( 0.42215817594807f / 8.0), FL2FXCONST_SGL(-0.07712787385208f / 8.0) },
+ { FL2FXCONST_SGL( 0.02704554141885f / 8.0), FL2FXCONST_SGL(-0.05417518053666f / 8.0) },
+ { FL2FXCONST_SGL( 0.80001773566818f / 8.0), FL2FXCONST_SGL( 0.91542195141039f / 8.0) },
+ { FL2FXCONST_SGL(-0.79351832348816f / 8.0), FL2FXCONST_SGL(-0.36208897989136f / 8.0) },
+ { FL2FXCONST_SGL( 0.63872359151636f / 8.0), FL2FXCONST_SGL( 0.08128252493444f / 8.0) },
+ { FL2FXCONST_SGL( 0.52890520960295f / 8.0), FL2FXCONST_SGL( 0.60048872455592f / 8.0) },
+ { FL2FXCONST_SGL( 0.74238552914587f / 8.0), FL2FXCONST_SGL( 0.04491915291044f / 8.0) },
+ { FL2FXCONST_SGL( 0.99096131449250f / 8.0), FL2FXCONST_SGL(-0.19451182854402f / 8.0) },
+ { FL2FXCONST_SGL(-0.80412329643109f / 8.0), FL2FXCONST_SGL(-0.88513818199457f / 8.0) },
+ { FL2FXCONST_SGL(-0.64612616129736f / 8.0), FL2FXCONST_SGL( 0.72198674804544f / 8.0) },
+ { FL2FXCONST_SGL( 0.11657770663191f / 8.0), FL2FXCONST_SGL(-0.83662833815041f / 8.0) },
+ { FL2FXCONST_SGL(-0.95053182488101f / 8.0), FL2FXCONST_SGL(-0.96939905138082f / 8.0) },
+ { FL2FXCONST_SGL(-0.62228872928622f / 8.0), FL2FXCONST_SGL( 0.82767262846661f / 8.0) },
+ { FL2FXCONST_SGL( 0.03004475787316f / 8.0), FL2FXCONST_SGL(-0.99738896333384f / 8.0) },
+ { FL2FXCONST_SGL(-0.97987214341034f / 8.0), FL2FXCONST_SGL( 0.36526129686425f / 8.0) },
+ { FL2FXCONST_SGL(-0.99986980746200f / 8.0), FL2FXCONST_SGL(-0.36021610299715f / 8.0) },
+ { FL2FXCONST_SGL( 0.89110648599879f / 8.0), FL2FXCONST_SGL(-0.97894250343044f / 8.0) },
+ { FL2FXCONST_SGL( 0.10407960510582f / 8.0), FL2FXCONST_SGL( 0.77357793811619f / 8.0) },
+ { FL2FXCONST_SGL( 0.95964737821728f / 8.0), FL2FXCONST_SGL(-0.35435818285502f / 8.0) },
+ { FL2FXCONST_SGL( 0.50843233159162f / 8.0), FL2FXCONST_SGL( 0.96107691266205f / 8.0) },
+ { FL2FXCONST_SGL( 0.17006334670615f / 8.0), FL2FXCONST_SGL(-0.76854025314829f / 8.0) },
+ { FL2FXCONST_SGL( 0.25872675063360f / 8.0), FL2FXCONST_SGL( 0.99893303933816f / 8.0) },
+ { FL2FXCONST_SGL(-0.01115998681937f / 8.0), FL2FXCONST_SGL( 0.98496019742444f / 8.0) },
+ { FL2FXCONST_SGL(-0.79598702973261f / 8.0), FL2FXCONST_SGL( 0.97138411318894f / 8.0) },
+ { FL2FXCONST_SGL(-0.99264708948101f / 8.0), FL2FXCONST_SGL(-0.99542822402536f / 8.0) },
+ { FL2FXCONST_SGL(-0.99829663752818f / 8.0), FL2FXCONST_SGL( 0.01877138824311f / 8.0) },
+ { FL2FXCONST_SGL(-0.70801016548184f / 8.0), FL2FXCONST_SGL( 0.33680685948117f / 8.0) },
+ { FL2FXCONST_SGL(-0.70467057786826f / 8.0), FL2FXCONST_SGL( 0.93272777501857f / 8.0) },
+ { FL2FXCONST_SGL( 0.99846021905254f / 8.0), FL2FXCONST_SGL(-0.98725746254433f / 8.0) },
+ { FL2FXCONST_SGL(-0.63364968534650f / 8.0), FL2FXCONST_SGL(-0.16473594423746f / 8.0) },
+ { FL2FXCONST_SGL(-0.16258217500792f / 8.0), FL2FXCONST_SGL(-0.95939125400802f / 8.0) },
+ { FL2FXCONST_SGL(-0.43645594360633f / 8.0), FL2FXCONST_SGL(-0.94805030113284f / 8.0) },
+ { FL2FXCONST_SGL(-0.99848471702976f / 8.0), FL2FXCONST_SGL( 0.96245166923809f / 8.0) },
+ { FL2FXCONST_SGL(-0.16796458968998f / 8.0), FL2FXCONST_SGL(-0.98987511890470f / 8.0) },
+ { FL2FXCONST_SGL(-0.87979225745213f / 8.0), FL2FXCONST_SGL(-0.71725725041680f / 8.0) },
+ { FL2FXCONST_SGL( 0.44183099021786f / 8.0), FL2FXCONST_SGL(-0.93568974498761f / 8.0) },
+ { FL2FXCONST_SGL( 0.93310180125532f / 8.0), FL2FXCONST_SGL(-0.99913308068246f / 8.0) },
+ { FL2FXCONST_SGL(-0.93941931782002f / 8.0), FL2FXCONST_SGL(-0.56409379640356f / 8.0) },
+ { FL2FXCONST_SGL(-0.88590003188677f / 8.0), FL2FXCONST_SGL( 0.47624600491382f / 8.0) },
+ { FL2FXCONST_SGL( 0.99971463703691f / 8.0), FL2FXCONST_SGL(-0.83889954253462f / 8.0) },
+ { FL2FXCONST_SGL(-0.75376385639978f / 8.0), FL2FXCONST_SGL( 0.00814643438625f / 8.0) },
+ { FL2FXCONST_SGL( 0.93887685615875f / 8.0), FL2FXCONST_SGL(-0.11284528204636f / 8.0) },
+ { FL2FXCONST_SGL( 0.85126435782309f / 8.0), FL2FXCONST_SGL( 0.52349251543547f / 8.0) },
+ { FL2FXCONST_SGL( 0.39701421446381f / 8.0), FL2FXCONST_SGL( 0.81779634174316f / 8.0) },
+ { FL2FXCONST_SGL(-0.37024464187437f / 8.0), FL2FXCONST_SGL(-0.87071656222959f / 8.0) },
+ { FL2FXCONST_SGL(-0.36024828242896f / 8.0), FL2FXCONST_SGL( 0.34655735648287f / 8.0) },
+ { FL2FXCONST_SGL(-0.93388812549209f / 8.0), FL2FXCONST_SGL(-0.84476541096429f / 8.0) },
+ { FL2FXCONST_SGL(-0.65298804552119f / 8.0), FL2FXCONST_SGL(-0.18439575450921f / 8.0) },
+ { FL2FXCONST_SGL( 0.11960319006843f / 8.0), FL2FXCONST_SGL( 0.99899346780168f / 8.0) },
+ { FL2FXCONST_SGL( 0.94292565553160f / 8.0), FL2FXCONST_SGL( 0.83163906518293f / 8.0) },
+ { FL2FXCONST_SGL( 0.75081145286948f / 8.0), FL2FXCONST_SGL(-0.35533223142265f / 8.0) },
+ { FL2FXCONST_SGL( 0.56721979748394f / 8.0), FL2FXCONST_SGL(-0.24076836414499f / 8.0) },
+ { FL2FXCONST_SGL( 0.46857766746029f / 8.0), FL2FXCONST_SGL(-0.30140233457198f / 8.0) },
+ { FL2FXCONST_SGL( 0.97312313923635f / 8.0), FL2FXCONST_SGL(-0.99548191630031f / 8.0) },
+ { FL2FXCONST_SGL(-0.38299976567017f / 8.0), FL2FXCONST_SGL( 0.98516909715427f / 8.0) },
+ { FL2FXCONST_SGL( 0.41025800019463f / 8.0), FL2FXCONST_SGL( 0.02116736935734f / 8.0) },
+ { FL2FXCONST_SGL( 0.09638062008048f / 8.0), FL2FXCONST_SGL( 0.04411984381457f / 8.0) },
+ { FL2FXCONST_SGL(-0.85283249275397f / 8.0), FL2FXCONST_SGL( 0.91475563922421f / 8.0) },
+ { FL2FXCONST_SGL( 0.88866808958124f / 8.0), FL2FXCONST_SGL(-0.99735267083226f / 8.0) },
+ { FL2FXCONST_SGL(-0.48202429536989f / 8.0), FL2FXCONST_SGL(-0.96805608884164f / 8.0) },
+ { FL2FXCONST_SGL( 0.27572582416567f / 8.0), FL2FXCONST_SGL( 0.58634753335832f / 8.0) },
+ { FL2FXCONST_SGL(-0.65889129659168f / 8.0), FL2FXCONST_SGL( 0.58835634138583f / 8.0) },
+ { FL2FXCONST_SGL( 0.98838086953732f / 8.0), FL2FXCONST_SGL( 0.99994349600236f / 8.0) },
+ { FL2FXCONST_SGL(-0.20651349620689f / 8.0), FL2FXCONST_SGL( 0.54593044066355f / 8.0) },
+ { FL2FXCONST_SGL(-0.62126416356920f / 8.0), FL2FXCONST_SGL(-0.59893681700392f / 8.0) },
+ { FL2FXCONST_SGL( 0.20320105410437f / 8.0), FL2FXCONST_SGL(-0.86879180355289f / 8.0) },
+ { FL2FXCONST_SGL(-0.97790548600584f / 8.0), FL2FXCONST_SGL( 0.96290806999242f / 8.0) },
+ { FL2FXCONST_SGL( 0.11112534735126f / 8.0), FL2FXCONST_SGL( 0.21484763313301f / 8.0) },
+ { FL2FXCONST_SGL(-0.41368337314182f / 8.0), FL2FXCONST_SGL( 0.28216837680365f / 8.0) },
+ { FL2FXCONST_SGL( 0.24133038992960f / 8.0), FL2FXCONST_SGL( 0.51294362630238f / 8.0) },
+ { FL2FXCONST_SGL(-0.66393410674885f / 8.0), FL2FXCONST_SGL(-0.08249679629081f / 8.0) },
+ { FL2FXCONST_SGL(-0.53697829178752f / 8.0), FL2FXCONST_SGL(-0.97649903936228f / 8.0) },
+ { FL2FXCONST_SGL(-0.97224737889348f / 8.0), FL2FXCONST_SGL( 0.22081333579837f / 8.0) },
+ { FL2FXCONST_SGL( 0.87392477144549f / 8.0), FL2FXCONST_SGL(-0.12796173740361f / 8.0) },
+ { FL2FXCONST_SGL( 0.19050361015753f / 8.0), FL2FXCONST_SGL( 0.01602615387195f / 8.0) },
+ { FL2FXCONST_SGL(-0.46353441212724f / 8.0), FL2FXCONST_SGL(-0.95249041539006f / 8.0) },
+ { FL2FXCONST_SGL(-0.07064096339021f / 8.0), FL2FXCONST_SGL(-0.94479803205886f / 8.0) },
+ { FL2FXCONST_SGL(-0.92444085484466f / 8.0), FL2FXCONST_SGL(-0.10457590187436f / 8.0) },
+ { FL2FXCONST_SGL(-0.83822593578728f / 8.0), FL2FXCONST_SGL(-0.01695043208885f / 8.0) },
+ { FL2FXCONST_SGL( 0.75214681811150f / 8.0), FL2FXCONST_SGL(-0.99955681042665f / 8.0) },
+ { FL2FXCONST_SGL(-0.42102998829339f / 8.0), FL2FXCONST_SGL( 0.99720941999394f / 8.0) },
+ { FL2FXCONST_SGL(-0.72094786237696f / 8.0), FL2FXCONST_SGL(-0.35008961934255f / 8.0) },
+ { FL2FXCONST_SGL( 0.78843311019251f / 8.0), FL2FXCONST_SGL( 0.52851398958271f / 8.0) },
+ { FL2FXCONST_SGL( 0.97394027897442f / 8.0), FL2FXCONST_SGL(-0.26695944086561f / 8.0) },
+ { FL2FXCONST_SGL( 0.99206463477946f / 8.0), FL2FXCONST_SGL(-0.57010120849429f / 8.0) },
+ { FL2FXCONST_SGL( 0.76789609461795f / 8.0), FL2FXCONST_SGL(-0.76519356730966f / 8.0) },
+ { FL2FXCONST_SGL(-0.82002421836409f / 8.0), FL2FXCONST_SGL(-0.73530179553767f / 8.0) },
+ { FL2FXCONST_SGL( 0.81924990025724f / 8.0), FL2FXCONST_SGL( 0.99698425250579f / 8.0) },
+ { FL2FXCONST_SGL(-0.26719850873357f / 8.0), FL2FXCONST_SGL( 0.68903369776193f / 8.0) },
+ { FL2FXCONST_SGL(-0.43311260380975f / 8.0), FL2FXCONST_SGL( 0.85321815947490f / 8.0) },
+ { FL2FXCONST_SGL( 0.99194979673836f / 8.0), FL2FXCONST_SGL( 0.91876249766422f / 8.0) },
+ { FL2FXCONST_SGL(-0.80692001248487f / 8.0), FL2FXCONST_SGL(-0.32627540663214f / 8.0) },
+ { FL2FXCONST_SGL( 0.43080003649976f / 8.0), FL2FXCONST_SGL(-0.21919095636638f / 8.0) },
+ { FL2FXCONST_SGL( 0.67709491937357f / 8.0), FL2FXCONST_SGL(-0.95478075822906f / 8.0) },
+ { FL2FXCONST_SGL( 0.56151770568316f / 8.0), FL2FXCONST_SGL(-0.70693811747778f / 8.0) },
+ { FL2FXCONST_SGL( 0.10831862810749f / 8.0), FL2FXCONST_SGL(-0.08628837174592f / 8.0) },
+ { FL2FXCONST_SGL( 0.91229417540436f / 8.0), FL2FXCONST_SGL(-0.65987351408410f / 8.0) },
+ { FL2FXCONST_SGL(-0.48972893932274f / 8.0), FL2FXCONST_SGL( 0.56289246362686f / 8.0) },
+ { FL2FXCONST_SGL(-0.89033658689697f / 8.0), FL2FXCONST_SGL(-0.71656563987082f / 8.0) },
+ { FL2FXCONST_SGL( 0.65269447475094f / 8.0), FL2FXCONST_SGL( 0.65916004833932f / 8.0) },
+ { FL2FXCONST_SGL( 0.67439478141121f / 8.0), FL2FXCONST_SGL(-0.81684380846796f / 8.0) },
+ { FL2FXCONST_SGL(-0.47770832416973f / 8.0), FL2FXCONST_SGL(-0.16789556203025f / 8.0) },
+ { FL2FXCONST_SGL(-0.99715979260878f / 8.0), FL2FXCONST_SGL(-0.93565784007648f / 8.0) },
+ { FL2FXCONST_SGL(-0.90889593602546f / 8.0), FL2FXCONST_SGL( 0.62034397054380f / 8.0) },
+ { FL2FXCONST_SGL(-0.06618622548177f / 8.0), FL2FXCONST_SGL(-0.23812217221359f / 8.0) },
+ { FL2FXCONST_SGL( 0.99430266919728f / 8.0), FL2FXCONST_SGL( 0.18812555317553f / 8.0) },
+ { FL2FXCONST_SGL( 0.97686402381843f / 8.0), FL2FXCONST_SGL(-0.28664534366620f / 8.0) },
+ { FL2FXCONST_SGL( 0.94813650221268f / 8.0), FL2FXCONST_SGL(-0.97506640027128f / 8.0) },
+ { FL2FXCONST_SGL(-0.95434497492853f / 8.0), FL2FXCONST_SGL(-0.79607978501983f / 8.0) },
+ { FL2FXCONST_SGL(-0.49104783137150f / 8.0), FL2FXCONST_SGL( 0.32895214359663f / 8.0) },
+ { FL2FXCONST_SGL( 0.99881175120751f / 8.0), FL2FXCONST_SGL( 0.88993983831354f / 8.0) },
+ { FL2FXCONST_SGL( 0.50449166760303f / 8.0), FL2FXCONST_SGL(-0.85995072408434f / 8.0) },
+ { FL2FXCONST_SGL( 0.47162891065108f / 8.0), FL2FXCONST_SGL(-0.18680204049569f / 8.0) },
+ { FL2FXCONST_SGL(-0.62081581361840f / 8.0), FL2FXCONST_SGL( 0.75000676218956f / 8.0) },
+ { FL2FXCONST_SGL(-0.43867015250812f / 8.0), FL2FXCONST_SGL( 0.99998069244322f / 8.0) },
+ { FL2FXCONST_SGL( 0.98630563232075f / 8.0), FL2FXCONST_SGL(-0.53578899600662f / 8.0) },
+ { FL2FXCONST_SGL(-0.61510362277374f / 8.0), FL2FXCONST_SGL(-0.89515019899997f / 8.0) },
+ { FL2FXCONST_SGL(-0.03841517601843f / 8.0), FL2FXCONST_SGL(-0.69888815681179f / 8.0) },
+ { FL2FXCONST_SGL(-0.30102157304644f / 8.0), FL2FXCONST_SGL(-0.07667808922205f / 8.0) },
+ { FL2FXCONST_SGL( 0.41881284182683f / 8.0), FL2FXCONST_SGL( 0.02188098922282f / 8.0) },
+ { FL2FXCONST_SGL(-0.86135454941237f / 8.0), FL2FXCONST_SGL( 0.98947480909359f / 8.0) },
+ { FL2FXCONST_SGL( 0.67226861393788f / 8.0), FL2FXCONST_SGL(-0.13494389011014f / 8.0) },
+ { FL2FXCONST_SGL(-0.70737398842068f / 8.0), FL2FXCONST_SGL(-0.76547349325992f / 8.0) },
+ { FL2FXCONST_SGL( 0.94044946687963f / 8.0), FL2FXCONST_SGL( 0.09026201157416f / 8.0) },
+ { FL2FXCONST_SGL(-0.82386352534327f / 8.0), FL2FXCONST_SGL( 0.08924768823676f / 8.0) },
+ { FL2FXCONST_SGL(-0.32070666698656f / 8.0), FL2FXCONST_SGL( 0.50143421908753f / 8.0) },
+ { FL2FXCONST_SGL( 0.57593163224487f / 8.0), FL2FXCONST_SGL(-0.98966422921509f / 8.0) },
+ { FL2FXCONST_SGL(-0.36326018419965f / 8.0), FL2FXCONST_SGL( 0.07440243123228f / 8.0) },
+ { FL2FXCONST_SGL( 0.99979044674350f / 8.0), FL2FXCONST_SGL(-0.14130287347405f / 8.0) },
+ { FL2FXCONST_SGL(-0.92366023326932f / 8.0), FL2FXCONST_SGL(-0.97979298068180f / 8.0) },
+ { FL2FXCONST_SGL(-0.44607178518598f / 8.0), FL2FXCONST_SGL(-0.54233252016394f / 8.0) },
+ { FL2FXCONST_SGL( 0.44226800932956f / 8.0), FL2FXCONST_SGL( 0.71326756742752f / 8.0) },
+ { FL2FXCONST_SGL( 0.03671907158312f / 8.0), FL2FXCONST_SGL( 0.63606389366675f / 8.0) },
+ { FL2FXCONST_SGL( 0.52175424682195f / 8.0), FL2FXCONST_SGL(-0.85396826735705f / 8.0) },
+ { FL2FXCONST_SGL(-0.94701139690956f / 8.0), FL2FXCONST_SGL(-0.01826348194255f / 8.0) },
+ { FL2FXCONST_SGL(-0.98759606946049f / 8.0), FL2FXCONST_SGL( 0.82288714303073f / 8.0) },
+ { FL2FXCONST_SGL( 0.87434794743625f / 8.0), FL2FXCONST_SGL( 0.89399495655433f / 8.0) },
+ { FL2FXCONST_SGL(-0.93412041758744f / 8.0), FL2FXCONST_SGL( 0.41374052024363f / 8.0) },
+ { FL2FXCONST_SGL( 0.96063943315511f / 8.0), FL2FXCONST_SGL( 0.93116709541280f / 8.0) },
+ { FL2FXCONST_SGL( 0.97534253457837f / 8.0), FL2FXCONST_SGL( 0.86150930812689f / 8.0) },
+ { FL2FXCONST_SGL( 0.99642466504163f / 8.0), FL2FXCONST_SGL( 0.70190043427512f / 8.0) },
+ { FL2FXCONST_SGL(-0.94705089665984f / 8.0), FL2FXCONST_SGL(-0.29580042814306f / 8.0) },
+ { FL2FXCONST_SGL( 0.91599807087376f / 8.0), FL2FXCONST_SGL(-0.98147830385781f / 8.0) }
+};
+//@}
+
+/*
+static const FIXP_SGL harmonicPhase [2][4] = {
+ { 1.0, 0.0, -1.0, 0.0},
+ { 0.0, 1.0, 0.0, -1.0}
+};
+*/
+
+
+/* The CLDFB-80 is not linear phase (unsymmetric), but the exact
+ phase difference between adjacent bands, at exact positions
+ (in this case exactly in the frequency band centre), can of
+ course be determined anyway. While the standard symmetric QMF
+ bank has a phase difference of 0.5*pi, the CLDFB-80
+ bank has the difference 0.2337*pi. */
+const FIXP_SGL harmonicPhaseX [2][4] = {
+ { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159475e-001),
+ FL2FXCONST_SGL(-7.423735494778152e-001), FL2FXCONST_SGL( 6.699862036159474e-001) },
+ { FL2FXCONST_SGL( 7.423735494778151e-001), FL2FXCONST_SGL( 6.699862036159476e-001),
+ FL2FXCONST_SGL(-7.423735494778151e-001), FL2FXCONST_SGL(-6.699862036159476e-001) }
+};
+
+/* tables for SBR and AAC LD */
+/* table for 8 time slot index */
+const int FDK_sbrDecoder_envelopeTable_8 [8][5] = {
+/* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */
+/* borders from left to right side; -1 = not in use */
+ /*[|T-|------]*/ { 2, 0, 0, 1, -1 },
+ /*[|-T-|-----]*/ { 2, 0, 0, 2, -1 },
+ /*[--|T-|----]*/ { 3, 1, 1, 2, 4 },
+ /*[---|T-|---]*/ { 3, 1, 1, 3, 5 },
+ /*[----|T-|--]*/ { 3, 1, 1, 4, 6 },
+ /*[-----|T--|]*/ { 2, 1, 1, 5, -1 },
+ /*[------|T-|]*/ { 2, 1, 1, 6, -1 },
+ /*[-------|T|]*/ { 2, 1, 1, 7, -1 },
+};
+
+/* table for 15 time slot index */
+const int FDK_sbrDecoder_envelopeTable_15 [15][6] = {
+ /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */
+ /* length from left to right side; -1 = not in use */
+ /*[|T---|------------]*/ { 2, 0, 0, 4, -1, -1},
+ /*[|-T---|-----------]*/ { 2, 0, 0, 5, -1, -1},
+ /*[|--|T---|---------]*/ { 3, 1, 1, 2, 6, -1},
+ /*[|---|T---|--------]*/ { 3, 1, 1, 3, 7, -1},
+ /*[|----|T---|-------]*/ { 3, 1, 1, 4, 8, -1},
+ /*[|-----|T---|------]*/ { 3, 1, 1, 5, 9, -1},
+ /*[|------|T---|-----]*/ { 3, 1, 1, 6, 10, -1},
+ /*[|-------|T---|----]*/ { 3, 1, 1, 7, 11, -1},
+ /*[|--------|T---|---]*/ { 3, 1, 1, 8, 12, -1},
+ /*[|---------|T---|--]*/ { 3, 1, 1, 9, 13, -1},
+ /*[|----------|T----|]*/ { 2, 1, 1,10, -1, -1},
+ /*[|-----------|T---|]*/ { 2, 1, 1,11, -1, -1},
+ /*[|------------|T--|]*/ { 2, 1, 1,12, -1, -1},
+ /*[|-------------|T-|]*/ { 2, 1, 1,13, -1, -1},
+ /*[|--------------|T|]*/ { 2, 1, 1,14, -1, -1},
+};
+
+/* table for 16 time slot index */
+const int FDK_sbrDecoder_envelopeTable_16 [16][6] = {
+ /* transientIndex nEnv, tranIdx, shortEnv, border1, border2, ... */
+ /* length from left to right side; -1 = not in use */
+ /*[|T---|------------|]*/ { 2, 0, 0, 4, -1, -1},
+ /*[|-T---|-----------|]*/ { 2, 0, 0, 5, -1, -1},
+ /*[|--|T---|----------]*/ { 3, 1, 1, 2, 6, -1},
+ /*[|---|T---|---------]*/ { 3, 1, 1, 3, 7, -1},
+ /*[|----|T---|--------]*/ { 3, 1, 1, 4, 8, -1},
+ /*[|-----|T---|-------]*/ { 3, 1, 1, 5, 9, -1},
+ /*[|------|T---|------]*/ { 3, 1, 1, 6, 10, -1},
+ /*[|-------|T---|-----]*/ { 3, 1, 1, 7, 11, -1},
+ /*[|--------|T---|----]*/ { 3, 1, 1, 8, 12, -1},
+ /*[|---------|T---|---]*/ { 3, 1, 1, 9, 13, -1},
+ /*[|----------|T---|--]*/ { 3, 1, 1,10, 14, -1},
+ /*[|-----------|T----|]*/ { 2, 1, 1,11, -1, -1},
+ /*[|------------|T---|]*/ { 2, 1, 1,12, -1, -1},
+ /*[|-------------|T--|]*/ { 2, 1, 1,13, -1, -1},
+ /*[|--------------|T-|]*/ { 2, 1, 1,14, -1, -1},
+ /*[|---------------|T|]*/ { 2, 1, 1,15, -1, -1},
+};
+
+/*!
+ \name FrameInfoDefaults
+
+ Predefined envelope positions for the FIX-FIX case (static framing)
+*/
+//@{
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15 = { 0, 1, {0, 15, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 15, 0} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15 = { 0, 2, {0, 8, 15, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 15} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15 = { 0, 4, {0, 4, 8, 12, 15, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 15} };
+#if (MAX_ENVELOPES >= 8)
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 15}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 15} };
+#endif
+
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16 = { 0, 1, {0, 16, 0, 0, 0, 0}, {1, 0, 0, 0, 0}, -1, 1, {0, 16, 0} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16 = { 0, 2, {0, 8, 16, 0, 0, 0}, {1, 1, 0, 0, 0}, -1, 2, {0, 8, 16} };
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16 = { 0, 4, {0, 4, 8, 12, 16, 0}, {1, 1, 1, 1, 0}, -1, 2, {0, 8, 16} };
+
+#if (MAX_ENVELOPES >= 8)
+const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16 = { 0, 8, {0, 2, 4, 6, 8, 10, 12, 14, 16}, {1, 1, 1, 1, 1, 1, 1, 1}, -1, 2, {0, 8, 16} };
+#endif
+
+
+//@}
+
+/*!
+ \name SBR_HuffmanTables
+
+ SBR Huffman Table Overview: \n
+ \n
+ o envelope level, 1.5 dB: \n
+ 1) sbr_huffBook_EnvLevel10T[120][2] \n
+ 2) sbr_huffBook_EnvLevel10F[120][2] \n
+ \n
+ o envelope balance, 1.5 dB: \n
+ 3) sbr_huffBook_EnvBalance10T[48][2] \n
+ 4) sbr_huffBook_EnvBalance10F[48][2] \n
+ \n
+ o envelope level, 3.0 dB: \n
+ 5) sbr_huffBook_EnvLevel11T[62][2] \n
+ 6) sbr_huffBook_EnvLevel11F[62][2] \n
+ \n
+ o envelope balance, 3.0 dB: \n
+ 7) sbr_huffBook_EnvBalance11T[24][2] \n
+ 8) sbr_huffBook_EnvBalance11F[24][2] \n
+ \n
+ o noise level, 3.0 dB: \n
+ 9) sbr_huffBook_NoiseLevel11T[62][2] \n
+ -) (sbr_huffBook_EnvLevel11F[62][2] is used for freq dir)\n
+ \n
+ o noise balance, 3.0 dB: \n
+ 10) sbr_huffBook_NoiseBalance11T[24][2]\n
+ -) (sbr_huffBook_EnvBalance11F[24][2] is used for freq dir)\n
+ \n
+ (1.5 dB is never used for noise)
+
+*/
+//@{
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2] = {
+ { 1, 2 }, { -64, -65 }, { 3, 4 }, { -63, -66 },
+ { 5, 6 }, { -62, -67 }, { 7, 8 }, { -61, -68 },
+ { 9, 10 }, { -60, -69 }, { 11, 12 }, { -59, -70 },
+ { 13, 14 }, { -58, -71 }, { 15, 16 }, { -57, -72 },
+ { 17, 18 }, { -73, -56 }, { 19, 21 }, { -74, 20 },
+ { -55, -75 }, { 22, 26 }, { 23, 24 }, { -54, -76 },
+ { -77, 25 }, { -53, -78 }, { 27, 34 }, { 28, 29 },
+ { -52, -79 }, { 30, 31 }, { -80, -51 }, { 32, 33 },
+ { -83, -82 }, { -81, -50 }, { 35, 57 }, { 36, 40 },
+ { 37, 38 }, { -88, -84 }, { -48, 39 }, { -90, -85 },
+ { 41, 46 }, { 42, 43 }, { -49, -87 }, { 44, 45 },
+ { -89, -86 }, {-124,-123 }, { 47, 50 }, { 48, 49 },
+ {-122,-121 }, {-120,-119 }, { 51, 54 }, { 52, 53 },
+ {-118,-117 }, {-116,-115 }, { 55, 56 }, {-114,-113 },
+ {-112,-111 }, { 58, 89 }, { 59, 74 }, { 60, 67 },
+ { 61, 64 }, { 62, 63 }, {-110,-109 }, {-108,-107 },
+ { 65, 66 }, {-106,-105 }, {-104,-103 }, { 68, 71 },
+ { 69, 70 }, {-102,-101 }, {-100, -99 }, { 72, 73 },
+ { -98, -97 }, { -96, -95 }, { 75, 82 }, { 76, 79 },
+ { 77, 78 }, { -94, -93 }, { -92, -91 }, { 80, 81 },
+ { -47, -46 }, { -45, -44 }, { 83, 86 }, { 84, 85 },
+ { -43, -42 }, { -41, -40 }, { 87, 88 }, { -39, -38 },
+ { -37, -36 }, { 90, 105 }, { 91, 98 }, { 92, 95 },
+ { 93, 94 }, { -35, -34 }, { -33, -32 }, { 96, 97 },
+ { -31, -30 }, { -29, -28 }, { 99, 102 }, { 100, 101 },
+ { -27, -26 }, { -25, -24 }, { 103, 104 }, { -23, -22 },
+ { -21, -20 }, { 106, 113 }, { 107, 110 }, { 108, 109 },
+ { -19, -18 }, { -17, -16 }, { 111, 112 }, { -15, -14 },
+ { -13, -12 }, { 114, 117 }, { 115, 116 }, { -11, -10 },
+ { -9, -8 }, { 118, 119 }, { -7, -6 }, { -5, -4 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2] = {
+ { 1, 2 }, { -64, -65 }, { 3, 4 }, { -63, -66 },
+ { 5, 6 }, { -67, -62 }, { 7, 8 }, { -68, -61 },
+ { 9, 10 }, { -69, -60 }, { 11, 13 }, { -70, 12 },
+ { -59, -71 }, { 14, 16 }, { -58, 15 }, { -72, -57 },
+ { 17, 19 }, { -73, 18 }, { -56, -74 }, { 20, 23 },
+ { 21, 22 }, { -55, -75 }, { -54, -53 }, { 24, 27 },
+ { 25, 26 }, { -76, -52 }, { -77, -51 }, { 28, 31 },
+ { 29, 30 }, { -50, -78 }, { -79, -49 }, { 32, 36 },
+ { 33, 34 }, { -48, -47 }, { -80, 35 }, { -81, -82 },
+ { 37, 47 }, { 38, 41 }, { 39, 40 }, { -83, -46 },
+ { -45, -84 }, { 42, 44 }, { -85, 43 }, { -44, -43 },
+ { 45, 46 }, { -88, -87 }, { -86, -90 }, { 48, 66 },
+ { 49, 56 }, { 50, 53 }, { 51, 52 }, { -92, -42 },
+ { -41, -39 }, { 54, 55 }, {-105, -89 }, { -38, -37 },
+ { 57, 60 }, { 58, 59 }, { -94, -91 }, { -40, -36 },
+ { 61, 63 }, { -20, 62 }, {-115,-110 }, { 64, 65 },
+ {-108,-107 }, {-101, -97 }, { 67, 89 }, { 68, 75 },
+ { 69, 72 }, { 70, 71 }, { -95, -93 }, { -34, -27 },
+ { 73, 74 }, { -22, -17 }, { -16,-124 }, { 76, 82 },
+ { 77, 79 }, {-123, 78 }, {-122,-121 }, { 80, 81 },
+ {-120,-119 }, {-118,-117 }, { 83, 86 }, { 84, 85 },
+ {-116,-114 }, {-113,-112 }, { 87, 88 }, {-111,-109 },
+ {-106,-104 }, { 90, 105 }, { 91, 98 }, { 92, 95 },
+ { 93, 94 }, {-103,-102 }, {-100, -99 }, { 96, 97 },
+ { -98, -96 }, { -35, -33 }, { 99, 102 }, { 100, 101 },
+ { -32, -31 }, { -30, -29 }, { 103, 104 }, { -28, -26 },
+ { -25, -24 }, { 106, 113 }, { 107, 110 }, { 108, 109 },
+ { -23, -21 }, { -19, -18 }, { 111, 112 }, { -15, -14 },
+ { -13, -12 }, { 114, 117 }, { 115, 116 }, { -11, -10 },
+ { -9, -8 }, { 118, 119 }, { -7, -6 }, { -5, -4 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2] = {
+ { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 },
+ { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 },
+ { -68, 9 }, { 10, 11 }, { -69, -59 }, { 12, 13 },
+ { -70, -58 }, { 14, 28 }, { 15, 21 }, { 16, 18 },
+ { -57, 17 }, { -71, -56 }, { 19, 20 }, { -88, -87 },
+ { -86, -85 }, { 22, 25 }, { 23, 24 }, { -84, -83 },
+ { -82, -81 }, { 26, 27 }, { -80, -79 }, { -78, -77 },
+ { 29, 36 }, { 30, 33 }, { 31, 32 }, { -76, -75 },
+ { -74, -73 }, { 34, 35 }, { -72, -55 }, { -54, -53 },
+ { 37, 41 }, { 38, 39 }, { -52, -51 }, { -50, 40 },
+ { -49, -48 }, { 42, 45 }, { 43, 44 }, { -47, -46 },
+ { -45, -44 }, { 46, 47 }, { -43, -42 }, { -41, -40 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 },
+ { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 },
+ { -60, 9 }, { 10, 11 }, { -69, -59 }, { -70, 12 },
+ { -58, 13 }, { 14, 17 }, { -71, 15 }, { -57, 16 },
+ { -56, -73 }, { 18, 32 }, { 19, 25 }, { 20, 22 },
+ { -72, 21 }, { -88, -87 }, { 23, 24 }, { -86, -85 },
+ { -84, -83 }, { 26, 29 }, { 27, 28 }, { -82, -81 },
+ { -80, -79 }, { 30, 31 }, { -78, -77 }, { -76, -75 },
+ { 33, 40 }, { 34, 37 }, { 35, 36 }, { -74, -55 },
+ { -54, -53 }, { 38, 39 }, { -52, -51 }, { -50, -49 },
+ { 41, 44 }, { 42, 43 }, { -48, -47 }, { -46, -45 },
+ { 45, 46 }, { -44, -43 }, { -42, 47 }, { -41, -40 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 },
+ { -62, 5 }, { -67, 6 }, { -61, 7 }, { -68, 8 },
+ { -60, 9 }, { 10, 11 }, { -69, -59 }, { 12, 14 },
+ { -70, 13 }, { -71, -58 }, { 15, 18 }, { 16, 17 },
+ { -72, -57 }, { -73, -74 }, { 19, 22 }, { -56, 20 },
+ { -55, 21 }, { -54, -77 }, { 23, 31 }, { 24, 25 },
+ { -75, -76 }, { 26, 27 }, { -78, -53 }, { 28, 29 },
+ { -52, -95 }, { -94, 30 }, { -93, -92 }, { 32, 47 },
+ { 33, 40 }, { 34, 37 }, { 35, 36 }, { -91, -90 },
+ { -89, -88 }, { 38, 39 }, { -87, -86 }, { -85, -84 },
+ { 41, 44 }, { 42, 43 }, { -83, -82 }, { -81, -80 },
+ { 45, 46 }, { -79, -51 }, { -50, -49 }, { 48, 55 },
+ { 49, 52 }, { 50, 51 }, { -48, -47 }, { -46, -45 },
+ { 53, 54 }, { -44, -43 }, { -42, -41 }, { 56, 59 },
+ { 57, 58 }, { -40, -39 }, { -38, -37 }, { 60, 61 },
+ { -36, -35 }, { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 },
+ { -62, 5 }, { -67, 6 }, { 7, 8 }, { -61, -68 },
+ { 9, 10 }, { -60, -69 }, { 11, 12 }, { -59, -70 },
+ { 13, 14 }, { -58, -71 }, { 15, 16 }, { -57, -72 },
+ { 17, 19 }, { -56, 18 }, { -55, -73 }, { 20, 24 },
+ { 21, 22 }, { -74, -54 }, { -53, 23 }, { -75, -76 },
+ { 25, 30 }, { 26, 27 }, { -52, -51 }, { 28, 29 },
+ { -77, -79 }, { -50, -49 }, { 31, 39 }, { 32, 35 },
+ { 33, 34 }, { -78, -46 }, { -82, -88 }, { 36, 37 },
+ { -83, -48 }, { -47, 38 }, { -86, -85 }, { 40, 47 },
+ { 41, 44 }, { 42, 43 }, { -80, -44 }, { -43, -42 },
+ { 45, 46 }, { -39, -87 }, { -84, -40 }, { 48, 55 },
+ { 49, 52 }, { 50, 51 }, { -95, -94 }, { -93, -92 },
+ { 53, 54 }, { -91, -90 }, { -89, -81 }, { 56, 59 },
+ { 57, 58 }, { -45, -41 }, { -38, -37 }, { 60, 61 },
+ { -36, -35 }, { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2] = {
+ { -64, 1 }, { -63, 2 }, { -65, 3 }, { -66, 4 },
+ { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 },
+ { -60, 9 }, { 10, 16 }, { 11, 13 }, { -69, 12 },
+ { -76, -75 }, { 14, 15 }, { -74, -73 }, { -72, -71 },
+ { 17, 20 }, { 18, 19 }, { -70, -59 }, { -58, -57 },
+ { 21, 22 }, { -56, -55 }, { -54, 23 }, { -53, -52 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 },
+ { -62, 5 }, { -61, 6 }, { -67, 7 }, { -68, 8 },
+ { -60, 9 }, { 10, 13 }, { -69, 11 }, { -59, 12 },
+ { -58, -76 }, { 14, 17 }, { 15, 16 }, { -75, -74 },
+ { -73, -72 }, { 18, 21 }, { 19, 20 }, { -71, -70 },
+ { -57, -56 }, { 22, 23 }, { -55, -54 }, { -53, -52 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2] = {
+ { -64, 1 }, { -63, 2 }, { -65, 3 }, { -66, 4 },
+ { -62, 5 }, { -67, 6 }, { 7, 8 }, { -61, -68 },
+ { 9, 30 }, { 10, 15 }, { -60, 11 }, { -69, 12 },
+ { 13, 14 }, { -59, -53 }, { -95, -94 }, { 16, 23 },
+ { 17, 20 }, { 18, 19 }, { -93, -92 }, { -91, -90 },
+ { 21, 22 }, { -89, -88 }, { -87, -86 }, { 24, 27 },
+ { 25, 26 }, { -85, -84 }, { -83, -82 }, { 28, 29 },
+ { -81, -80 }, { -79, -78 }, { 31, 46 }, { 32, 39 },
+ { 33, 36 }, { 34, 35 }, { -77, -76 }, { -75, -74 },
+ { 37, 38 }, { -73, -72 }, { -71, -70 }, { 40, 43 },
+ { 41, 42 }, { -58, -57 }, { -56, -55 }, { 44, 45 },
+ { -54, -52 }, { -51, -50 }, { 47, 54 }, { 48, 51 },
+ { 49, 50 }, { -49, -48 }, { -47, -46 }, { 52, 53 },
+ { -45, -44 }, { -43, -42 }, { 55, 58 }, { 56, 57 },
+ { -41, -40 }, { -39, -38 }, { 59, 60 }, { -37, -36 },
+ { -35, 61 }, { -34, -33 }
+};
+
+const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { 4, 9 },
+ { -66, 5 }, { -62, 6 }, { 7, 8 }, { -76, -75 },
+ { -74, -73 }, { 10, 17 }, { 11, 14 }, { 12, 13 },
+ { -72, -71 }, { -70, -69 }, { 15, 16 }, { -68, -67 },
+ { -61, -60 }, { 18, 21 }, { 19, 20 }, { -59, -58 },
+ { -57, -56 }, { 22, 23 }, { -55, -54 }, { -53, -52 }
+};
+//@}
+
+
+
+
+/*!
+ \name parametric stereo
+ \brief constants used by the parametric stereo part of the decoder
+
+*/
+
+
+/* constants used in psbitdec.cpp */
+
+/* FIX_BORDER can have 0, 1, 2, 4 envelopes */
+const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4] = {0, 1, 2, 4};
+
+
+/* IID & ICC Huffman codebooks */
+const SCHAR aBookPsIidTimeDecode[28][2] = {
+ { -64, 1 }, { -65, 2 }, { -63, 3 }, { -66, 4 },
+ { -62, 5 }, { -67, 6 }, { -61, 7 }, { -68, 8 },
+ { -60, 9 }, { -69, 10 }, { -59, 11 }, { -70, 12 },
+ { -58, 13 }, { -57, 14 }, { -71, 15 }, { 16, 17 },
+ { -56, -72 }, { 18, 21 }, { 19, 20 }, { -55, -78 },
+ { -77, -76 }, { 22, 25 }, { 23, 24 }, { -75, -74 },
+ { -73, -54 }, { 26, 27 }, { -53, -52 }, { -51, -50 }
+};
+
+const SCHAR aBookPsIidFreqDecode[28][2] = {
+ { -64, 1 }, { 2, 3 }, { -63, -65 }, { 4, 5 },
+ { -62, -66 }, { 6, 7 }, { -61, -67 }, { 8, 9 },
+ { -68, -60 }, { -59, 10 }, { -69, 11 }, { -58, 12 },
+ { -70, 13 }, { -71, 14 }, { -57, 15 }, { 16, 17 },
+ { -56, -72 }, { 18, 19 }, { -55, -54 }, { 20, 21 },
+ { -73, -53 }, { 22, 24 }, { -74, 23 }, { -75, -78 },
+ { 25, 26 }, { -77, -76 }, { -52, 27 }, { -51, -50 }
+};
+
+const SCHAR aBookPsIccTimeDecode[14][2] = {
+ { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 },
+ { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 },
+ { -68, 9 }, { -59, 10 }, { -69, 11 }, { -58, 12 },
+ { -70, 13 }, { -71, -57 }
+};
+
+const SCHAR aBookPsIccFreqDecode[14][2] = {
+ { -64, 1 }, { -63, 2 }, { -65, 3 }, { -62, 4 },
+ { -66, 5 }, { -61, 6 }, { -67, 7 }, { -60, 8 },
+ { -59, 9 }, { -68, 10 }, { -58, 11 }, { -69, 12 },
+ { -57, 13 }, { -70, -71 }
+};
+
+/* IID-fine Huffman codebooks */
+
+const SCHAR aBookPsIidFineTimeDecode[60][2] = {
+ { 1, -64 }, { -63, 2 }, { 3, -65 }, { 4, 59 },
+ { 5, 7 }, { 6, -67 }, { -68, -60 }, { -61, 8 },
+ { 9, 11 }, { -59, 10 }, { -70, -58 }, { 12, 41 },
+ { 13, 20 }, { 14, -71 }, { -55, 15 }, { -53, 16 },
+ { 17, -77 }, { 18, 19 }, { -85, -84 }, { -46, -45 },
+ { -57, 21 }, { 22, 40 }, { 23, 29 }, { -51, 24 },
+ { 25, 26 }, { -83, -82 }, { 27, 28 }, { -90, -38 },
+ { -92, -91 }, { 30, 37 }, { 31, 34 }, { 32, 33 },
+ { -35, -34 }, { -37, -36 }, { 35, 36 }, { -94, -93 },
+ { -89, -39 }, { 38, -79 }, { 39, -81 }, { -88, -40 },
+ { -74, -54 }, { 42, -69 }, { 43, 44 }, { -72, -56 },
+ { 45, 52 }, { 46, 50 }, { 47, -76 }, { -49, 48 },
+ { -47, 49 }, { -87, -41 }, { -52, 51 }, { -78, -50 },
+ { 53, -73 }, { 54, -75 }, { 55, 57 }, { 56, -80 },
+ { -86, -42 }, { -48, 58 }, { -44, -43 }, { -66, -62 }
+};
+
+
+const SCHAR aBookPsIidFineFreqDecode[60][2] = {
+ { 1, -64 }, { 2, 4 }, { 3, -65 }, { -66, -62 },
+ { -63, 5 }, { 6, 7 }, { -67, -61 }, { 8, 9 },
+ { -68, -60 }, { 10, 11 }, { -69, -59 }, { 12, 13 },
+ { -70, -58 }, { 14, 18 }, { -57, 15 }, { 16, -72 },
+ { -54, 17 }, { -75, -53 }, { 19, 37 }, { -56, 20 },
+ { 21, -73 }, { 22, 29 }, { 23, -76 }, { 24, -78 },
+ { 25, 28 }, { 26, 27 }, { -85, -43 }, { -83, -45 },
+ { -81, -47 }, { -52, 30 }, { -50, 31 }, { 32, -79 },
+ { 33, 34 }, { -82, -46 }, { 35, 36 }, { -90, -89 },
+ { -92, -91 }, { 38, -71 }, { -55, 39 }, { 40, -74 },
+ { 41, 50 }, { 42, -77 }, { -49, 43 }, { 44, 47 },
+ { 45, 46 }, { -86, -42 }, { -88, -87 }, { 48, 49 },
+ { -39, -38 }, { -41, -40 }, { -51, 51 }, { 52, 59 },
+ { 53, 56 }, { 54, 55 }, { -35, -34 }, { -37, -36 },
+ { 57, 58 }, { -94, -93 }, { -84, -44 }, { -80, -48 }
+};
+
+/* constants used in psdec.cpp */
+
+const FIXP_DBL decayScaleFactTable[64] = {
+
+ FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000), FL2FXCONST_DBL(1.000000),
+ FL2FXCONST_DBL(0.950000), FL2FXCONST_DBL(0.900000), FL2FXCONST_DBL(0.850000), FL2FXCONST_DBL(0.800000),
+ FL2FXCONST_DBL(0.750000), FL2FXCONST_DBL(0.700000), FL2FXCONST_DBL(0.650000), FL2FXCONST_DBL(0.600000),
+ FL2FXCONST_DBL(0.550000), FL2FXCONST_DBL(0.500000), FL2FXCONST_DBL(0.450000), FL2FXCONST_DBL(0.400000),
+ FL2FXCONST_DBL(0.350000), FL2FXCONST_DBL(0.300000), FL2FXCONST_DBL(0.250000), FL2FXCONST_DBL(0.200000),
+ FL2FXCONST_DBL(0.150000), FL2FXCONST_DBL(0.100000), FL2FXCONST_DBL(0.050000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000),
+ FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000), FL2FXCONST_DBL(0.000000) };
+
+/* the values of the following 3 tables are shiftet right by 1 ! */
+const FIXP_DBL ScaleFactors[NO_IID_LEVELS] = {
+
+ 0x5a5ded00, 0x59cd0400, 0x58c29680, 0x564c2e80, 0x52a3d480,
+ 0x4c8be080, 0x46df3080, 0x40000000, 0x384ba5c0, 0x304c2980,
+ 0x24e9f640, 0x1b4a2940, 0x11b5c0a0, 0x0b4e2540, 0x0514ea90
+};
+
+const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE] = {
+
+ 0x5a825c00, 0x5a821c00, 0x5a815100, 0x5a7ed000, 0x5a76e600,
+ 0x5a5ded00, 0x5a39b880, 0x59f1fd00, 0x5964d680, 0x5852ca00,
+ 0x564c2e80, 0x54174480, 0x50ea7500, 0x4c8be080, 0x46df3080,
+ 0x40000000, 0x384ba5c0, 0x304c2980, 0x288dd240, 0x217a2900,
+ 0x1b4a2940, 0x13c5ece0, 0x0e2b0090, 0x0a178ef0, 0x072ab798,
+ 0x0514ea90, 0x02dc5944, 0x019bf87c, 0x00e7b173, 0x00824b8b,
+ 0x00494568
+};
+const FIXP_DBL Alphas[NO_ICC_LEVELS] = {
+
+ 0x00000000, 0x0b6b5be0, 0x12485f80, 0x1da2fa40,
+ 0x2637ebc0, 0x3243f6c0, 0x466b7480, 0x6487ed80
+};
+
+#if defined(ARCH_PREFER_MULT_32x16)
+#define FIXP_PS FIXP_SGL
+#define FXP_CAST(a) FX_DBL2FX_SGL((FIXP_DBL)a)
+#define FL2FXCONST_PS FL2FXCONST_SGL
+#else
+#define FIXP_PS FIXP_DBL
+#define FXP_CAST
+#define FL2FXCONST_PS FL2FXCONST_DBL
+#endif
+
+const FIXP_PS aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS] = {
+FXP_CAST(0x53625b00), FXP_CAST(0x4848af00), FXP_CAST(0x3ea94d00) };
+
+const FIXP_PS aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS] = {
+FXP_CAST(0x68b92180), FXP_CAST(0xde396900), FXP_CAST(0x80650380), FXP_CAST(0xcb537e40), FXP_CAST(0x5beb8f00), FXP_CAST(0x72f29200), FXP_CAST(0xf1f43c50), FXP_CAST(0x83896280),
+FXP_CAST(0xb9b99c00), FXP_CAST(0x4cda8f00), FXP_CAST(0x7a576e00), FXP_CAST(0x060799e0), FXP_CAST(0x89be5280), FXP_CAST(0xa9dab600), FXP_CAST(0x3be51b00), FXP_CAST(0x7eb91900),
+FXP_CAST(0x19f4f540), FXP_CAST(0x92dcb380), FXP_CAST(0x9c1ad700), FXP_CAST(0x29761940), FXP_CAST(0x7ffbf500), FXP_CAST(0x2d3eb180), FXP_CAST(0x9eab0a00), FXP_CAST(0x90d0aa80),
+FXP_CAST(0x1601bcc0), FXP_CAST(0x7e180e80), FXP_CAST(0x3f6b3940), FXP_CAST(0xacdeeb00), FXP_CAST(0x88435b00), FXP_CAST(0x0202a768), FXP_CAST(0x79194f80), FXP_CAST(0x5007fd00),
+FXP_CAST(0xbd1ecf00), FXP_CAST(0x82a8d100), FXP_CAST(0xedf6e5e0), FXP_CAST(0x711f3500), FXP_CAST(0x5eac4480), FXP_CAST(0xcf0447c0), FXP_CAST(0x80245f80), FXP_CAST(0xda5cd4c0),
+FXP_CAST(0x665c0800), FXP_CAST(0x6afbc500), FXP_CAST(0xe21e85e0), FXP_CAST(0x80c5e500), FXP_CAST(0xc7b003c0), FXP_CAST(0x59139f80), FXP_CAST(0x74a8e400), FXP_CAST(0xf5f51f40),
+FXP_CAST(0x84896680), FXP_CAST(0xb6662b00), FXP_CAST(0x4999b600), FXP_CAST(0x7b76a300), FXP_CAST(0x0a0b0650), FXP_CAST(0x8b572b80), FXP_CAST(0xa6ec4580), FXP_CAST(0x384fda80),
+FXP_CAST(0x7f3a1f00), FXP_CAST(0x1de19ec0), FXP_CAST(0x95045000), FXP_CAST(0x99a3e180), FXP_CAST(0x25a30740), FXP_CAST(0x7fdb9e80), FXP_CAST(0x30fbdb00), FXP_CAST(0xa153d500) };
+
+const FIXP_PS aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS] = {
+FXP_CAST(0xb6663a80), FXP_CAST(0x84896200), FXP_CAST(0xf5f50c70), FXP_CAST(0x74a8dc80), FXP_CAST(0x5913ad00), FXP_CAST(0xc7b01480), FXP_CAST(0x80c5e300), FXP_CAST(0xe21e73a0),
+FXP_CAST(0x6afbba80), FXP_CAST(0x665c1380), FXP_CAST(0xda5ce6c0), FXP_CAST(0x80246080), FXP_CAST(0xcf043640), FXP_CAST(0x5eac3800), FXP_CAST(0x711f3e00), FXP_CAST(0xedf6f8a0),
+FXP_CAST(0x82a8d500), FXP_CAST(0xbd1ebe80), FXP_CAST(0x5007ee00), FXP_CAST(0x79195580), FXP_CAST(0x0202ba40), FXP_CAST(0x88436180), FXP_CAST(0xacdedc80), FXP_CAST(0x3f6b28c0),
+FXP_CAST(0x7e181180), FXP_CAST(0x1601cf40), FXP_CAST(0x90d0b380), FXP_CAST(0x9eaafd80), FXP_CAST(0x2d3e9fc0), FXP_CAST(0x7ffbf580), FXP_CAST(0x29762b00), FXP_CAST(0x9c1ae280),
+FXP_CAST(0x92dca980), FXP_CAST(0x19f4e2c0), FXP_CAST(0x7eb91680), FXP_CAST(0x3be52b80), FXP_CAST(0xa9dac400), FXP_CAST(0x89be4b80), FXP_CAST(0x06078710), FXP_CAST(0x7a576880),
+FXP_CAST(0x4cda9e00), FXP_CAST(0xb9b9ac00), FXP_CAST(0x83895e00), FXP_CAST(0xf1f42990), FXP_CAST(0x72f28a00), FXP_CAST(0x5beb9c00), FXP_CAST(0xcb538f40), FXP_CAST(0x80650200),
+FXP_CAST(0xde3956c0), FXP_CAST(0x68b91680), FXP_CAST(0x68b92c00), FXP_CAST(0xde397b40), FXP_CAST(0x80650500), FXP_CAST(0xcb536d00), FXP_CAST(0x5beb8180), FXP_CAST(0x72f29a80),
+FXP_CAST(0xf1f44f10), FXP_CAST(0x83896700), FXP_CAST(0xb9b98c80), FXP_CAST(0x4cda8000), FXP_CAST(0x7a577380), FXP_CAST(0x0607acb8), FXP_CAST(0x89be5a00), FXP_CAST(0xa9daa800) };
+
+const FIXP_PS aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS] = {
+FXP_CAST(0x7e807380), FXP_CAST(0x72b9bb00), FXP_CAST(0x5c44ee80), FXP_CAST(0x3d3938c0), FXP_CAST(0x80000000), FXP_CAST(0x80000000),
+FXP_CAST(0x72b9bb00), FXP_CAST(0x7e807380), FXP_CAST(0xba914700), FXP_CAST(0x050677b0), FXP_CAST(0x895cc380), FXP_CAST(0x834e4900) };
+
+const FIXP_PS aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS] = {
+FXP_CAST(0xec791720), FXP_CAST(0xc73ca080), FXP_CAST(0xa748ea00), FXP_CAST(0x8f976980), FXP_CAST(0x00000000), FXP_CAST(0x00000000),
+FXP_CAST(0x38c35f80), FXP_CAST(0x1386e8e0), FXP_CAST(0x9477d000), FXP_CAST(0x80194380), FXP_CAST(0xcff26140), FXP_CAST(0x1ce70d40) };
+
+const FIXP_PS aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0x63e52480), FXP_CAST(0x30fbc540), FXP_CAST(0x6d73af00)}, {FXP_CAST(0xc7b01280), FXP_CAST(0x89be5100), FXP_CAST(0xf7c31cb0)}, {FXP_CAST(0x83896200), FXP_CAST(0x7641af00), FXP_CAST(0x8aee2700)},
+{FXP_CAST(0x0202b330), FXP_CAST(0xcf043ac0), FXP_CAST(0x9bfab500)}, {FXP_CAST(0x7d572c80), FXP_CAST(0xcf043ac0), FXP_CAST(0x1893b960)}, {FXP_CAST(0x34ac7fc0), FXP_CAST(0x7641af00), FXP_CAST(0x7abf7980)},
+{FXP_CAST(0x99a3ee00), FXP_CAST(0x89be5100), FXP_CAST(0x58eead80)}, {FXP_CAST(0x9eab0580), FXP_CAST(0x30fbc540), FXP_CAST(0xd77dae40)}, {FXP_CAST(0x3be52140), FXP_CAST(0x30fbc540), FXP_CAST(0x819b8500)},
+{FXP_CAST(0x7b769d80), FXP_CAST(0x89be5100), FXP_CAST(0xb3a12280)}, {FXP_CAST(0xf9f86878), FXP_CAST(0x7641af00), FXP_CAST(0x37c519c0)}, {FXP_CAST(0x81e7ef80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7ff16880)},
+{FXP_CAST(0xcf043cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x3e8b2340)}, {FXP_CAST(0x68b92280), FXP_CAST(0x7641af00), FXP_CAST(0xb9e4a900)}, {FXP_CAST(0x5eac3980), FXP_CAST(0x89be5100), FXP_CAST(0x80a05200)},
+{FXP_CAST(0xc094cd00), FXP_CAST(0x30fbc540), FXP_CAST(0xd051dc80)}, {FXP_CAST(0x85a89400), FXP_CAST(0x30fbc540), FXP_CAST(0x53483b00)}, {FXP_CAST(0x0a0af5e0), FXP_CAST(0x89be5100), FXP_CAST(0x7cb1b680)},
+{FXP_CAST(0x7eb91900), FXP_CAST(0x7641af00), FXP_CAST(0x2006e8c0)}, {FXP_CAST(0x2d3ea680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa0ec1c00)}, {FXP_CAST(0x95044180), FXP_CAST(0xcf043ac0), FXP_CAST(0x880d2180)},
+{FXP_CAST(0xa4147300), FXP_CAST(0x7641af00), FXP_CAST(0xf0282870)}, {FXP_CAST(0x42e13f80), FXP_CAST(0x89be5100), FXP_CAST(0x694c4a00)}, {FXP_CAST(0x79195200), FXP_CAST(0x30fbc540), FXP_CAST(0x71374780)},
+{FXP_CAST(0xf1f43550), FXP_CAST(0x30fbc540), FXP_CAST(0xff6593ea)}, {FXP_CAST(0x80c5e280), FXP_CAST(0x89be5100), FXP_CAST(0x8e39ec00)}, {FXP_CAST(0xd689e480), FXP_CAST(0x7641af00), FXP_CAST(0x97648100)},
+{FXP_CAST(0x6d235300), FXP_CAST(0xcf043ac0), FXP_CAST(0x110a20c0)}, {FXP_CAST(0x5913a800), FXP_CAST(0xcf043ac0), FXP_CAST(0x785d4f80)}, {FXP_CAST(0xb9b99a00), FXP_CAST(0x7641af00), FXP_CAST(0x5e440880)},
+{FXP_CAST(0x88436100), FXP_CAST(0x89be5100), FXP_CAST(0xdece7000)}, {FXP_CAST(0x12091320), FXP_CAST(0x30fbc540), FXP_CAST(0x8309f800)}, {FXP_CAST(0x7f9afd00), FXP_CAST(0x30fbc540), FXP_CAST(0xada33f00)},
+{FXP_CAST(0x25a31700), FXP_CAST(0x89be5100), FXP_CAST(0x30cc3600)}, {FXP_CAST(0x90d0ab80), FXP_CAST(0x7641af00), FXP_CAST(0x7f7cbe80)}, {FXP_CAST(0xa9dabf00), FXP_CAST(0xcf043ac0), FXP_CAST(0x45182580)},
+{FXP_CAST(0x4999cb80), FXP_CAST(0xcf043ac0), FXP_CAST(0xc0681c80)}, {FXP_CAST(0x7641ac80), FXP_CAST(0x7641af00), FXP_CAST(0x80194380)}, {FXP_CAST(0xe9fe3300), FXP_CAST(0x89be5100), FXP_CAST(0xc95184c0)},
+{FXP_CAST(0x80246000), FXP_CAST(0x30fbc540), FXP_CAST(0x4d55d800)}, {FXP_CAST(0xde396fc0), FXP_CAST(0x30fbc540), FXP_CAST(0x7e324000)}, {FXP_CAST(0x711f3f00), FXP_CAST(0x89be5100), FXP_CAST(0x275ce480)},
+{FXP_CAST(0x53211700), FXP_CAST(0x7641af00), FXP_CAST(0xa6343580)}, {FXP_CAST(0xb3256780), FXP_CAST(0xcf043ac0), FXP_CAST(0x85997b80)}, {FXP_CAST(0x8b572680), FXP_CAST(0xcf043ac0), FXP_CAST(0xe89ba660)},
+{FXP_CAST(0x19f4f780), FXP_CAST(0x7641af00), FXP_CAST(0x64c4e100)}, {FXP_CAST(0x7ffbf580), FXP_CAST(0x89be5100), FXP_CAST(0x7493a380)}, {FXP_CAST(0x1de18100), FXP_CAST(0x30fbc540), FXP_CAST(0x070897f0)},
+{FXP_CAST(0x8d0d6a80), FXP_CAST(0x30fbc540), FXP_CAST(0x91ed6f00)}, {FXP_CAST(0xaff81380), FXP_CAST(0x89be5100), FXP_CAST(0x932db000)}, {FXP_CAST(0x5007fb00), FXP_CAST(0x7641af00), FXP_CAST(0x0970feb0)},
+{FXP_CAST(0x72f28d00), FXP_CAST(0xcf043ac0), FXP_CAST(0x758d6500)}, {FXP_CAST(0xe21e6cc0), FXP_CAST(0xcf043ac0), FXP_CAST(0x63436f80)}, {FXP_CAST(0x80040b00), FXP_CAST(0x7641af00), FXP_CAST(0xe63d7600)},
+{FXP_CAST(0xe60b1ae0), FXP_CAST(0x89be5100), FXP_CAST(0x84ea5c80)}, {FXP_CAST(0x74a8e100), FXP_CAST(0x30fbc540), FXP_CAST(0xa7f07500)}, {FXP_CAST(0x4cda8980), FXP_CAST(0x30fbc540), FXP_CAST(0x29a6d340)},
+{FXP_CAST(0xacdeda80), FXP_CAST(0x89be5100), FXP_CAST(0x7e93d600)}, {FXP_CAST(0x8ee0c980), FXP_CAST(0x7641af00), FXP_CAST(0x4b662680)}, {FXP_CAST(0x21c6a280), FXP_CAST(0xcf043ac0), FXP_CAST(0xc7258c80)},
+{FXP_CAST(0x7fdb9f00), FXP_CAST(0xcf043ac0), FXP_CAST(0x8006d500)}, {FXP_CAST(0x1601ba60), FXP_CAST(0x7641af00), FXP_CAST(0xc2830940)}, {FXP_CAST(0x89be4c80), FXP_CAST(0x89be5100), FXP_CAST(0x471cf100)},
+{FXP_CAST(0xb6664400), FXP_CAST(0x30fbc540), FXP_CAST(0x7f3fb800)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0xaff80c80), FXP_CAST(0x89be5100), FXP_CAST(0xbda29e00)}, {FXP_CAST(0x8d0d6f00), FXP_CAST(0x30fbc540), FXP_CAST(0x8043ee80)}, {FXP_CAST(0x1de18a20), FXP_CAST(0x30fbc540), FXP_CAST(0xcc3e7840)},
+{FXP_CAST(0x7ffbf500), FXP_CAST(0x89be5100), FXP_CAST(0x4fdfc180)}, {FXP_CAST(0x19f4ee40), FXP_CAST(0x7641af00), FXP_CAST(0x7d9e4c00)}, {FXP_CAST(0x8b572300), FXP_CAST(0xcf043ac0), FXP_CAST(0x244a2940)},
+{FXP_CAST(0xb3256f00), FXP_CAST(0xcf043ac0), FXP_CAST(0xa3f0a500)}, {FXP_CAST(0x53211e00), FXP_CAST(0x7641af00), FXP_CAST(0x86944500)}, {FXP_CAST(0x711f3a80), FXP_CAST(0x89be5100), FXP_CAST(0xebc72040)},
+{FXP_CAST(0xde3966c0), FXP_CAST(0x30fbc540), FXP_CAST(0x66b87e00)}, {FXP_CAST(0x80246080), FXP_CAST(0x30fbc540), FXP_CAST(0x73362c00)}, {FXP_CAST(0xe9fe3c40), FXP_CAST(0x89be5100), FXP_CAST(0x03d1d110)},
+{FXP_CAST(0x7641b000), FXP_CAST(0x7641af00), FXP_CAST(0x90520c80)}, {FXP_CAST(0x4999c380), FXP_CAST(0xcf043ac0), FXP_CAST(0x94e80a80)}, {FXP_CAST(0xa9dab800), FXP_CAST(0xcf043ac0), FXP_CAST(0x0ca570e0)},
+{FXP_CAST(0x90d0b000), FXP_CAST(0x7641af00), FXP_CAST(0x76c9bc80)}, {FXP_CAST(0x25a32000), FXP_CAST(0x89be5100), FXP_CAST(0x61338500)}, {FXP_CAST(0x7f9afc80), FXP_CAST(0x30fbc540), FXP_CAST(0xe318f060)},
+{FXP_CAST(0x120909c0), FXP_CAST(0x30fbc540), FXP_CAST(0x84124e00)}, {FXP_CAST(0x88435d80), FXP_CAST(0x89be5100), FXP_CAST(0xaa4d2f80)}, {FXP_CAST(0xb9b9a200), FXP_CAST(0x7641af00), FXP_CAST(0x2cae1800)},
+{FXP_CAST(0x5913ae80), FXP_CAST(0xcf043ac0), FXP_CAST(0x7f040680)}, {FXP_CAST(0x6d234e00), FXP_CAST(0xcf043ac0), FXP_CAST(0x48c6a100)}, {FXP_CAST(0xd689db80), FXP_CAST(0x7641af00), FXP_CAST(0xc44860c0)},
+{FXP_CAST(0x80c5e380), FXP_CAST(0x89be5100), FXP_CAST(0x80005d00)}, {FXP_CAST(0xf1f43eb0), FXP_CAST(0x30fbc540), FXP_CAST(0xc55a3a00)}, {FXP_CAST(0x79195500), FXP_CAST(0x30fbc540), FXP_CAST(0x49c3de00)},
+{FXP_CAST(0x42e13700), FXP_CAST(0x89be5100), FXP_CAST(0x7edc5b00)}, {FXP_CAST(0xa4146c80), FXP_CAST(0x7641af00), FXP_CAST(0x2b8c2c00)}, {FXP_CAST(0x95044680), FXP_CAST(0xcf043ac0), FXP_CAST(0xa968c100)},
+{FXP_CAST(0x2d3eaf40), FXP_CAST(0xcf043ac0), FXP_CAST(0x8460fd80)}, {FXP_CAST(0x7eb91780), FXP_CAST(0x7641af00), FXP_CAST(0xe44621e0)}, {FXP_CAST(0x0a0aec80), FXP_CAST(0x89be5100), FXP_CAST(0x61fb5c00)},
+{FXP_CAST(0x85a89100), FXP_CAST(0x30fbc540), FXP_CAST(0x76555780)}, {FXP_CAST(0xc094d500), FXP_CAST(0x30fbc540), FXP_CAST(0x0b71f790)}, {FXP_CAST(0x5eac4000), FXP_CAST(0x89be5100), FXP_CAST(0x94401a80)},
+{FXP_CAST(0x68b91d80), FXP_CAST(0x7641af00), FXP_CAST(0x90ea3980)}, {FXP_CAST(0xcf043440), FXP_CAST(0xcf043ac0), FXP_CAST(0x05067a08)}, {FXP_CAST(0x81e7f180), FXP_CAST(0xcf043ac0), FXP_CAST(0x73bb6d00)},
+{FXP_CAST(0xf9f871e0), FXP_CAST(0x7641af00), FXP_CAST(0x65ff0e00)}, {FXP_CAST(0x7b76a000), FXP_CAST(0x89be5100), FXP_CAST(0xea9664c0)}, {FXP_CAST(0x3be518c0), FXP_CAST(0x30fbc540), FXP_CAST(0x8633e880)},
+{FXP_CAST(0x9eaaff00), FXP_CAST(0x30fbc540), FXP_CAST(0xa4c84500)}, {FXP_CAST(0x99a3f400), FXP_CAST(0x89be5100), FXP_CAST(0x2571eac0)}, {FXP_CAST(0x34ac8840), FXP_CAST(0x7641af00), FXP_CAST(0x7dd82b00)},
+{FXP_CAST(0x7d572a80), FXP_CAST(0xcf043ac0), FXP_CAST(0x4eed8400)}, {FXP_CAST(0x0202a9c4), FXP_CAST(0xcf043ac0), FXP_CAST(0xcb249700)}, {FXP_CAST(0x83896000), FXP_CAST(0x7641af00), FXP_CAST(0x80318200)},
+{FXP_CAST(0xc7b01b00), FXP_CAST(0x89be5100), FXP_CAST(0xbeab7580)}, {FXP_CAST(0x63e52a80), FXP_CAST(0x30fbc540), FXP_CAST(0x4364b700)}, {FXP_CAST(0x63e51f00), FXP_CAST(0x30fbc540), FXP_CAST(0x7fa6bd00)},
+{FXP_CAST(0xc7b00a00), FXP_CAST(0x89be5100), FXP_CAST(0x32a67940)}, {FXP_CAST(0x83896400), FXP_CAST(0x7641af00), FXP_CAST(0xaf2fd200)}, {FXP_CAST(0x0202bc9c), FXP_CAST(0xcf043ac0), FXP_CAST(0x829e6e80)},
+{FXP_CAST(0x7d572e80), FXP_CAST(0xcf043ac0), FXP_CAST(0xdcde6b80)}, {FXP_CAST(0x34ac7700), FXP_CAST(0x7641af00), FXP_CAST(0x5ce4e280)}, {FXP_CAST(0x99a3e880), FXP_CAST(0x89be5100), FXP_CAST(0x79089c00)},
+{FXP_CAST(0x9eab0b80), FXP_CAST(0x30fbc540), FXP_CAST(0x1307ae80)}, {FXP_CAST(0x3be52980), FXP_CAST(0x30fbc540), FXP_CAST(0x98906880)}, {FXP_CAST(0x7b769b00), FXP_CAST(0x89be5100), FXP_CAST(0x8d51b300)},
+{FXP_CAST(0xf9f85f10), FXP_CAST(0x7641af00), FXP_CAST(0xfd62ee24)}, {FXP_CAST(0x81e7ee00), FXP_CAST(0xcf043ac0), FXP_CAST(0x70439680)}, {FXP_CAST(0xcf044580), FXP_CAST(0xcf043ac0), FXP_CAST(0x6a6d9600)},
+{FXP_CAST(0x68b92800), FXP_CAST(0x7641af00), FXP_CAST(0xf2275f80)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x55063900), FXP_CAST(0x0c8bd360), FXP_CAST(0x636c0400)},
+{FXP_CAST(0x3084ca00), FXP_CAST(0xc3a94580), FXP_CAST(0x4a0d6700)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)}, {FXP_CAST(0x80000000), FXP_CAST(0x80000000), FXP_CAST(0x80000000)},
+{FXP_CAST(0x6fec9a80), FXP_CAST(0x5133cc80), FXP_CAST(0x7573df00)}, {FXP_CAST(0x7e2df000), FXP_CAST(0x7a7d0580), FXP_CAST(0x7ed03e00)}, {FXP_CAST(0xa4c84280), FXP_CAST(0xb8e31300), FXP_CAST(0xd5af0140)},
+{FXP_CAST(0xf0f488a0), FXP_CAST(0x8275a100), FXP_CAST(0x1a72e360)}, {FXP_CAST(0x80aaa680), FXP_CAST(0x471ced00), FXP_CAST(0x9d2ead80)}, {FXP_CAST(0x9477d100), FXP_CAST(0x7d8a5f00), FXP_CAST(0x8151df80)}};
+
+const FIXP_PS aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS] = {
+{FXP_CAST(0xea7d08a0), FXP_CAST(0xdad7f3c0), FXP_CAST(0xee9c9f60)}, {FXP_CAST(0xc1e54140), FXP_CAST(0x9d0dfe80), FXP_CAST(0xcd1e7300)}, {FXP_CAST(0xa051a580), FXP_CAST(0x809dc980), FXP_CAST(0xaf61c400)},
+{FXP_CAST(0x898d4e00), FXP_CAST(0x8f1d3400), FXP_CAST(0x97988280)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)}, {FXP_CAST(0x00000000), FXP_CAST(0x00000000), FXP_CAST(0x00000000)},
+{FXP_CAST(0x3e1abec0), FXP_CAST(0x62f20180), FXP_CAST(0x32e18d00)}, {FXP_CAST(0x1582f760), FXP_CAST(0x25280c40), FXP_CAST(0x116360a0)}, {FXP_CAST(0xa6343800), FXP_CAST(0x6a6d9880), FXP_CAST(0x87327a00)},
+{FXP_CAST(0x80e32200), FXP_CAST(0xe70747c0), FXP_CAST(0x82c32b00)}, {FXP_CAST(0xf2f42420), FXP_CAST(0x6a6d9880), FXP_CAST(0xaea47080)}, {FXP_CAST(0x456eba00), FXP_CAST(0xe70747c0), FXP_CAST(0xedaa8640)}};
+
+const FIXP_PS p8_13_20[13] =
+{
+ FL2FXCONST_PS(0.00746082949812f), FL2FXCONST_PS(0.02270420949825f), FL2FXCONST_PS(0.04546865930473f), FL2FXCONST_PS(0.07266113929591f),
+ FL2FXCONST_PS(0.09885108575264f), FL2FXCONST_PS(0.11793710567217f), FL2FXCONST_PS(0.125f ), FL2FXCONST_PS(0.11793710567217f),
+ FL2FXCONST_PS(0.09885108575264f), FL2FXCONST_PS(0.07266113929591f), FL2FXCONST_PS(0.04546865930473f), FL2FXCONST_PS(0.02270420949825f),
+ FL2FXCONST_PS(0.00746082949812f)
+};
+
+const FIXP_PS p2_13_20[13] =
+{
+ FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f), FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f),
+ FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.30596630545168f), FL2FXCONST_PS(0.5f), FL2FXCONST_PS( 0.30596630545168f),
+ FL2FXCONST_PS(0.0f), FL2FXCONST_PS(-0.07293139167538f), FL2FXCONST_PS(0.0f), FL2FXCONST_PS( 0.01899487526049f),
+ FL2FXCONST_PS(0.0f)
+};
+
+
+
+const UCHAR aAllpassLinkDelaySer[] = { 3, 4, 5};
+
+const UCHAR delayIndexQmf[NO_QMF_CHANNELS] = {
+ 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14,
+ 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14,
+ 14, 14, 14, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1
+};
+
+const UCHAR groupBorders20[NO_IID_GROUPS + 1] =
+{
+ 6, 7, 0, 1, 2, 3, /* 6 subqmf subbands - 0th qmf subband */
+ 9, 8, /* 2 subqmf subbands - 1st qmf subband */
+ 10, 11, /* 2 subqmf subbands - 2nd qmf subband */
+ 3, 4, 5, 6, 7, 8,
+ 9, 11, 14, 18, 23, 35, 64
+};
+
+const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1] =
+{
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, /* 12 subqmf subbands - 0th qmf subband */
+ 12, 13, 14, 15, 16, 17, 18, 19, /* 8 subqmf subbands - 1st qmf subband */
+ 20, 21, 22, 23, /* 4 subqmf subbands - 2nd qmf subband */
+ 24, 25, 26, 27, /* 4 subqmf subbands - 3nd qmf subband */
+ 28, 29, 30, 31, /* 4 subqmf subbands - 4nd qmf subband */
+ 32-27, 33-27, 34-27, 35-27, 36-27, 37-27, 38-27,
+ 40-27, 42-27, 44-27, 46-27, 48-27, 51-27, 54-27,
+ 57-27, 60-27, 64-27, 68-27, 91-27
+};
+
+const UCHAR bins2groupMap20[NO_IID_GROUPS] =
+{
+ 1, 0,
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19
+};
+
+const UCHAR quantizedIIDs[NO_IID_STEPS] =
+{
+ 2, 4, 7, 10, 14, 18, 25
+};
+const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE] =
+{
+ 2, 4, 6, 8, 10, 13, 16, 19, 22, 25, 30, 35, 40, 45, 50
+};
+
+const UCHAR FDK_sbrDecoder_aNoIidBins[3] = {NO_LOW_RES_IID_BINS,
+ NO_MID_RES_IID_BINS,
+ NO_HI_RES_IID_BINS};
+
+const UCHAR FDK_sbrDecoder_aNoIccBins[3] = {NO_LOW_RES_ICC_BINS,
+ NO_MID_RES_ICC_BINS,
+ NO_HI_RES_ICC_BINS};
+
+
+
+/************************************************************************/
+/*!
+ \brief Create lookup tables for some arithmetic functions
+
+ The tables would normally be defined as const arrays,
+ but initialization at run time allows to specify their accuracy.
+*/
+/************************************************************************/
+
+/* 1/x-table: (example for INV_TABLE_BITS 8)
+
+ The table covers an input range from 0.5 to 1.0 with a step size of 1/512,
+ starting at 0.5 + 1/512.
+ Each table entry corresponds to an input interval starting 1/1024 below the
+ exact value and ending 1/1024 above it.
+
+ The table is actually a 0.5/x-table, so that the output range is again
+ 0.5...1.0 and the exponent of the result must be increased by 1.
+
+ Input range Index in table result
+ -------------------------------------------------------------------
+ 0.500000...0.500976 - 0.5 / 0.500000 = 1.000000
+ 0.500976...0.502930 0 0.5 / 0.501953 = 0.996109
+ 0.502930...0.500488 1 0.5 / 0.503906 = 0.992248
+ ...
+ 0.999023...1.000000 255 0.5 / 1.000000 = 0.500000
+
+ for (i=0; i<INV_TABLE_SIZE; i++) {
+ d = 0.5f / ( 0.5f+(double)(i+1)/(INV_TABLE_SIZE*2) ) ;
+ invTable[i] = FL2FX_SGL(d);
+ }
+*/
+const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE] =
+{
+ 0x7f80, 0x7f01, 0x7e83, 0x7e07, 0x7d8b, 0x7d11, 0x7c97, 0x7c1e,
+ 0x7ba6, 0x7b2f, 0x7ab9, 0x7a44, 0x79cf, 0x795c, 0x78e9, 0x7878,
+ 0x7807, 0x7796, 0x7727, 0x76b9, 0x764b, 0x75de, 0x7572, 0x7506,
+ 0x749c, 0x7432, 0x73c9, 0x7360, 0x72f9, 0x7292, 0x722c, 0x71c6,
+ 0x7161, 0x70fd, 0x709a, 0x7037, 0x6fd5, 0x6f74, 0x6f13, 0x6eb3,
+ 0x6e54, 0x6df5, 0x6d97, 0x6d39, 0x6cdc, 0x6c80, 0x6c24, 0x6bc9,
+ 0x6b6f, 0x6b15, 0x6abc, 0x6a63, 0x6a0b, 0x69b3, 0x695c, 0x6906,
+ 0x68b0, 0x685a, 0x6806, 0x67b1, 0x675e, 0x670a, 0x66b8, 0x6666,
+ 0x6614, 0x65c3, 0x6572, 0x6522, 0x64d2, 0x6483, 0x6434, 0x63e6,
+ 0x6399, 0x634b, 0x62fe, 0x62b2, 0x6266, 0x621b, 0x61d0, 0x6185,
+ 0x613b, 0x60f2, 0x60a8, 0x6060, 0x6017, 0x5fcf, 0x5f88, 0x5f41,
+ 0x5efa, 0x5eb4, 0x5e6e, 0x5e28, 0x5de3, 0x5d9f, 0x5d5a, 0x5d17,
+ 0x5cd3, 0x5c90, 0x5c4d, 0x5c0b, 0x5bc9, 0x5b87, 0x5b46, 0x5b05,
+ 0x5ac4, 0x5a84, 0x5a44, 0x5a05, 0x59c6, 0x5987, 0x5949, 0x590a,
+ 0x58cd, 0x588f, 0x5852, 0x5815, 0x57d9, 0x579d, 0x5761, 0x5725,
+ 0x56ea, 0x56af, 0x5675, 0x563b, 0x5601, 0x55c7, 0x558e, 0x5555,
+ 0x551c, 0x54e3, 0x54ab, 0x5473, 0x543c, 0x5405, 0x53ce, 0x5397,
+ 0x5360, 0x532a, 0x52f4, 0x52bf, 0x5289, 0x5254, 0x521f, 0x51eb,
+ 0x51b7, 0x5183, 0x514f, 0x511b, 0x50e8, 0x50b5, 0x5082, 0x5050,
+ 0x501d, 0x4feb, 0x4fba, 0x4f88, 0x4f57, 0x4f26, 0x4ef5, 0x4ec4,
+ 0x4e94, 0x4e64, 0x4e34, 0x4e04, 0x4dd5, 0x4da6, 0x4d77, 0x4d48,
+ 0x4d19, 0x4ceb, 0x4cbd, 0x4c8f, 0x4c61, 0x4c34, 0x4c07, 0x4bd9,
+ 0x4bad, 0x4b80, 0x4b54, 0x4b27, 0x4afb, 0x4acf, 0x4aa4, 0x4a78,
+ 0x4a4d, 0x4a22, 0x49f7, 0x49cd, 0x49a2, 0x4978, 0x494e, 0x4924,
+ 0x48fa, 0x48d1, 0x48a7, 0x487e, 0x4855, 0x482d, 0x4804, 0x47dc,
+ 0x47b3, 0x478b, 0x4763, 0x473c, 0x4714, 0x46ed, 0x46c5, 0x469e,
+ 0x4677, 0x4651, 0x462a, 0x4604, 0x45de, 0x45b8, 0x4592, 0x456c,
+ 0x4546, 0x4521, 0x44fc, 0x44d7, 0x44b2, 0x448d, 0x4468, 0x4444,
+ 0x441f, 0x43fb, 0x43d7, 0x43b3, 0x4390, 0x436c, 0x4349, 0x4325,
+ 0x4302, 0x42df, 0x42bc, 0x4299, 0x4277, 0x4254, 0x4232, 0x4210,
+ 0x41ee, 0x41cc, 0x41aa, 0x4189, 0x4167, 0x4146, 0x4125, 0x4104,
+ 0x40e3, 0x40c2, 0x40a1, 0x4081, 0x4060, 0x4040, 0x4020, 0x4000
+};
+
diff --git a/libSBRdec/src/sbr_rom.h b/libSBRdec/src/sbr_rom.h
new file mode 100644
index 0000000..912f7e4
--- /dev/null
+++ b/libSBRdec/src/sbr_rom.h
@@ -0,0 +1,232 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+\brief Declaration of constant tables
+
+*/
+#ifndef __rom_H
+#define __rom_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+#include "qmf.h"
+
+#define INV_INT_TABLE_SIZE 49
+#define SBR_NF_NO_RANDOM_VAL 512 /*!< Size of random number array for noise floor */
+
+/*
+ Frequency scales
+*/
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_16[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_22[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_24[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16];
+
+/*
+ Low-Power-Profile Transposer
+*/
+#define NUM_WHFACTOR_TABLE_ENTRIES 9
+extern const USHORT FDK_sbrDecoder_sbr_whFactorsIndex[NUM_WHFACTOR_TABLE_ENTRIES];
+extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRIES][6];
+
+
+
+/*
+ Envelope Adjustor
+*/
+extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4];
+extern const UCHAR FDK_sbrDecoder_sbr_limGains_e[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4];
+extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2];
+extern const FIXP_SGL harmonicPhaseX [2][4];
+
+/*
+ Envelope Extractor
+*/
+extern const int FDK_sbrDecoder_envelopeTable_8 [8][5];
+extern const int FDK_sbrDecoder_envelopeTable_15 [15][6];
+extern const int FDK_sbrDecoder_envelopeTable_16 [16][6];
+
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_15;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_15;
+
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info1_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info2_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info4_16;
+extern const FRAME_INFO FDK_sbrDecoder_sbr_frame_info8_16;
+
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10T[120][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel10F[120][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10T[48][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance10F[48][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11T[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvLevel11F[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11T[24][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_EnvBalance11F[24][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseLevel11T[62][2];
+extern const SCHAR FDK_sbrDecoder_sbr_huffBook_NoiseBalance11T[24][2];
+
+
+/*
+ Parametric stereo
+*/
+
+
+extern const FIXP_DBL decayScaleFactTable[NO_QMF_CHANNELS];
+
+/* FIX_BORDER can have 0, 1, 2, 4 envelops */
+extern const UCHAR FDK_sbrDecoder_aFixNoEnvDecode[4];
+
+/* IID & ICC Huffman codebooks */
+extern const SCHAR aBookPsIidTimeDecode[28][2];
+extern const SCHAR aBookPsIidFreqDecode[28][2];
+extern const SCHAR aBookPsIccTimeDecode[14][2];
+extern const SCHAR aBookPsIccFreqDecode[14][2];
+
+/* IID-fine Huffman codebooks */
+
+extern const SCHAR aBookPsIidFineTimeDecode[60][2];
+extern const SCHAR aBookPsIidFineFreqDecode[60][2];
+
+/* the values of the following 3 tables are shiftet right by 1 ! */
+extern const FIXP_DBL ScaleFactors[NO_IID_LEVELS];
+extern const FIXP_DBL ScaleFactorsFine[NO_IID_LEVELS_FINE];
+extern const FIXP_DBL Alphas[NO_ICC_LEVELS];
+
+#if defined(ARCH_PREFER_MULT_32x16)
+extern const FIXP_SGL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS];
+extern const FIXP_SGL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS];
+
+extern const FIXP_SGL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_SGL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+
+extern const FIXP_SGL p8_13_20[13];
+extern const FIXP_SGL p2_13_20[13];
+
+#else
+extern const FIXP_DBL aAllpassLinkDecaySer[NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorReQmf[NO_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorImQmf[NO_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorReSubQmf20[NO_SUB_QMF_CHANNELS];
+extern const FIXP_DBL aaFractDelayPhaseFactorImSubQmf20[NO_SUB_QMF_CHANNELS];
+
+extern const FIXP_DBL aaFractDelayPhaseFactorSerReQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerImQmf[NO_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerReSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+extern const FIXP_DBL aaFractDelayPhaseFactorSerImSubQmf20[NO_SUB_QMF_CHANNELS][NO_SERIAL_ALLPASS_LINKS];
+
+extern const FIXP_DBL p8_13_20[13];
+extern const FIXP_DBL p2_13_20[13];
+#endif
+
+extern const UCHAR aAllpassLinkDelaySer[3];
+extern const UCHAR delayIndexQmf[NO_QMF_CHANNELS];
+extern const UCHAR groupBorders20[NO_IID_GROUPS + 1];
+extern const UCHAR groupBorders34[NO_IID_GROUPS_HI_RES + 1];
+extern const UCHAR bins2groupMap20[NO_IID_GROUPS];
+extern const UCHAR quantizedIIDs[NO_IID_STEPS];
+extern const UCHAR quantizedIIDsFine[NO_IID_STEPS_FINE];
+extern const UCHAR FDK_sbrDecoder_aNoIidBins[3];
+extern const UCHAR FDK_sbrDecoder_aNoIccBins[3];
+
+
+/* Lookup tables for some arithmetic functions */
+
+#define INV_TABLE_BITS 8
+#define INV_TABLE_SIZE (1<<INV_TABLE_BITS)
+extern const FIXP_SGL FDK_sbrDecoder_invTable[INV_TABLE_SIZE];
+
+#endif // __rom_H
diff --git a/libSBRdec/src/sbr_scale.h b/libSBRdec/src/sbr_scale.h
new file mode 100644
index 0000000..a98fc05
--- /dev/null
+++ b/libSBRdec/src/sbr_scale.h
@@ -0,0 +1,123 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+\brief Sbr scaling factors,
+To deal with the dynamic range in the different processing stages, a
+fixed point specific code has to rely on scaling factors. A floating
+point code carries a scaling factor -- the exponent -- for each value,
+so scaling is not necessary there.
+
+The output of the core decoder (low band) is scaled up to cover as much
+as possible bits for each value. As high band and low band are processed
+in different algorithm sections, they require their own scaling
+factors. In addition, any static buffers, e.g. filter states, require a
+separate scaling factor as well. The code takes care to do the proper
+adjustment, if scaling factors of a filter state and the time signal differ.
+
+\sa #QMF_SCALE_FACTOR, \ref documentationOverview
+*/
+
+#ifndef __SBR_SCALE_H
+#define __SBR_SCALE_H
+
+/*!
+\verbatim
+ scale:
+ 0 left aligned e.g. |max| >=0.5
+ FRACT_BITS-1 zero e.g |max| = 0
+\endverbatim
+
+ Dynamic scaling is used to achieve sufficient accuracy even when the signal
+ energy is low. The dynamic framing of SBR produces a variable overlap area
+ where samples from the previous QMF-Analysis are stored. Depending on the
+ start position and stop position of the current SBR envelopes, the processing
+ buffer consists of differently scaled regions like illustrated in the below
+ figure.
+
+ \image html scales.png Scale
+*/
+
+
+#endif
diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp
new file mode 100644
index 0000000..ce5247a
--- /dev/null
+++ b/libSBRdec/src/sbrdec_drc.cpp
@@ -0,0 +1,512 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************** MPEG-4 AAC Decoder **************************
+
+ Author(s): Christian Griebel
+ Description: Dynamic range control (DRC) decoder tool for SBR
+
+******************************************************************************/
+
+#include "sbrdec_drc.h"
+
+
+/* DRC - Offset table for QMF interpolation. */
+static const int offsetTab[2][16] =
+{
+ { 0, 4, 8, 12, 16, 20, 24, 28, 0, 0, 0, 0, 0, 0, 0, 0 }, /* 1024 framing */
+ { 0, 4, 8, 12, 16, 19, 22, 26, 0, 0, 0, 0, 0, 0, 0, 0 } /* 960 framing */
+};
+
+/*!
+ \brief Initialize DRC QMF factors
+
+ \hDrcData Handle to DRC channel data.
+
+ \return none
+*/
+void sbrDecoder_drcInitChannel (
+ HANDLE_SBR_DRC_CHANNEL hDrcData )
+{
+ int band;
+
+ if (hDrcData == NULL) {
+ return;
+ }
+
+ for (band = 0; band < (64); band++) {
+ hDrcData->prevFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+ }
+
+ for (band = 0; band < SBRDEC_MAX_DRC_BANDS; band++) {
+ hDrcData->currFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+ hDrcData->nextFact_mag[band] = (FIXP_DBL)MAXVAL_DBL /*FL2FXCONST_DBL(1.0f)*/;
+ }
+
+ hDrcData->prevFact_exp = 0;
+ hDrcData->currFact_exp = 0;
+ hDrcData->nextFact_exp = 0;
+
+ hDrcData->numBandsCurr = 0;
+ hDrcData->numBandsNext = 0;
+
+ hDrcData->winSequenceCurr = 0;
+ hDrcData->winSequenceNext = 0;
+
+ hDrcData->drcInterpolationSchemeCurr = 0;
+ hDrcData->drcInterpolationSchemeNext = 0;
+
+ hDrcData->enable = 0;
+}
+
+
+/*!
+ \brief Swap DRC QMF scaling factors after they have been applied.
+
+ \hDrcData Handle to DRC channel data.
+
+ \return none
+*/
+void sbrDecoder_drcUpdateChannel (
+ HANDLE_SBR_DRC_CHANNEL hDrcData )
+{
+ if (hDrcData == NULL) {
+ return;
+ }
+ if (hDrcData->enable != 1) {
+ return;
+ }
+
+ /* swap previous data */
+ FDKmemcpy( hDrcData->currFact_mag,
+ hDrcData->nextFact_mag,
+ SBRDEC_MAX_DRC_BANDS * sizeof(FIXP_DBL) );
+
+ hDrcData->currFact_exp = hDrcData->nextFact_exp;
+
+ hDrcData->numBandsCurr = hDrcData->numBandsNext;
+
+ FDKmemcpy( hDrcData->bandTopCurr,
+ hDrcData->bandTopNext,
+ SBRDEC_MAX_DRC_BANDS * sizeof(USHORT) );
+
+ hDrcData->drcInterpolationSchemeCurr = hDrcData->drcInterpolationSchemeNext;
+
+ hDrcData->winSequenceCurr = hDrcData->winSequenceNext;
+}
+
+
+/*!
+ \brief Apply DRC factors slot based.
+
+ \hDrcData Handle to DRC channel data.
+ \qmfRealSlot Pointer to real valued QMF data of one time slot.
+ \qmfImagSlot Pointer to the imaginary QMF data of one time slot.
+ \col Number of the time slot.
+ \numQmfSubSamples Total number of time slots for one frame.
+ \scaleFactor Pointer to the out scale factor of the time slot.
+
+ \return None.
+*/
+void sbrDecoder_drcApplySlot (
+ HANDLE_SBR_DRC_CHANNEL hDrcData,
+ FIXP_DBL *qmfRealSlot,
+ FIXP_DBL *qmfImagSlot,
+ int col,
+ int numQmfSubSamples,
+ int maxShift
+ )
+{
+ const int *offset;
+
+ int band, bottomMdct, topMdct, bin, useLP;
+ int indx = numQmfSubSamples - (numQmfSubSamples >> 1) - 10; /* l_border */
+ int frameLenFlag = (numQmfSubSamples == 30) ? 1 : 0;
+
+ const FIXP_DBL *fact_mag = NULL;
+ INT fact_exp = 0;
+ UINT numBands = 0;
+ USHORT *bandTop = NULL;
+ int shortDrc = 0;
+
+ FIXP_DBL alphaValue = FL2FXCONST_DBL(0.0f);
+
+ if (hDrcData == NULL) {
+ return;
+ }
+ if (hDrcData->enable != 1) {
+ return;
+ }
+
+ offset = offsetTab[frameLenFlag];
+
+ useLP = (qmfImagSlot == NULL) ? 1 : 0;
+
+ col += indx;
+ bottomMdct = 0;
+ bin = 0;
+
+ /* get respective data and calc interpolation factor */
+ if (col < (numQmfSubSamples>>1)) { /* first half of current frame */
+ if (hDrcData->winSequenceCurr != 2) { /* long window */
+ int j = col + (numQmfSubSamples>>1);
+
+ if (hDrcData->drcInterpolationSchemeCurr == 0) {
+ INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+ alphaValue = (FIXP_DBL)(j * k);
+ }
+ else {
+ if (j >= offset[hDrcData->drcInterpolationSchemeCurr - 1]) {
+ alphaValue = FL2FXCONST_DBL(1.0f);
+ }
+ }
+ }
+ else { /* short windows */
+ shortDrc = 1;
+ }
+
+ fact_mag = hDrcData->currFact_mag;
+ fact_exp = hDrcData->currFact_exp;
+ numBands = hDrcData->numBandsCurr;
+ bandTop = hDrcData->bandTopCurr;
+ }
+ else if (col < numQmfSubSamples) { /* second half of current frame */
+ if (hDrcData->winSequenceNext != 2) { /* next: long window */
+ int j = col - (numQmfSubSamples>>1);
+
+ if (hDrcData->drcInterpolationSchemeNext == 0) {
+ INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+ alphaValue = (FIXP_DBL)(j * k);
+ }
+ else {
+ if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) {
+ alphaValue = FL2FXCONST_DBL(1.0f);
+ }
+ }
+
+ fact_mag = hDrcData->nextFact_mag;
+ fact_exp = hDrcData->nextFact_exp;
+ numBands = hDrcData->numBandsNext;
+ bandTop = hDrcData->bandTopNext;
+ }
+ else { /* next: short windows */
+ if (hDrcData->winSequenceCurr != 2) { /* current: long window */
+ alphaValue = (FIXP_DBL)0;
+
+ fact_mag = hDrcData->nextFact_mag;
+ fact_exp = hDrcData->nextFact_exp;
+ numBands = hDrcData->numBandsNext;
+ bandTop = hDrcData->bandTopNext;
+ }
+ else { /* current: short windows */
+ shortDrc = 1;
+
+ fact_mag = hDrcData->currFact_mag;
+ fact_exp = hDrcData->currFact_exp;
+ numBands = hDrcData->numBandsCurr;
+ bandTop = hDrcData->bandTopCurr;
+ }
+ }
+ }
+ else { /* first half of next frame */
+ if (hDrcData->winSequenceNext != 2) { /* long window */
+ int j = col - (numQmfSubSamples>>1);
+
+ if (hDrcData->drcInterpolationSchemeNext == 0) {
+ INT k = (frameLenFlag) ? 0x4444444 : 0x4000000;
+
+ alphaValue = (FIXP_DBL)(j * k);
+ }
+ else {
+ if (j >= offset[hDrcData->drcInterpolationSchemeNext - 1]) {
+ alphaValue = FL2FXCONST_DBL(1.0f);
+ }
+ }
+ }
+ else { /* short windows */
+ shortDrc = 1;
+ }
+
+ fact_mag = hDrcData->nextFact_mag;
+ fact_exp = hDrcData->nextFact_exp;
+ numBands = hDrcData->numBandsNext;
+ bandTop = hDrcData->bandTopNext;
+
+ col -= numQmfSubSamples;
+ }
+
+
+ /* process bands */
+ for (band = 0; band < (int)numBands; band++) {
+ int bottomQmf, topQmf;
+
+ FIXP_DBL drcFact_mag = FL2FXCONST_DBL(1.0f);
+
+ topMdct = (bandTop[band]+1) << 2;
+
+ if (!shortDrc) { /* long window */
+ if (frameLenFlag) {
+ /* 960 framing */
+ bottomMdct = 30 * (bottomMdct / 30);
+ topMdct = 30 * (topMdct / 30);
+
+ bottomQmf = fMultIfloor((FIXP_DBL)0x4444444, bottomMdct);
+ topQmf = fMultIfloor((FIXP_DBL)0x4444444, topMdct);
+ }
+ else {
+ /* 1024 framing */
+ bottomMdct &= ~0x1f;
+ topMdct &= ~0x1f;
+
+ bottomQmf = bottomMdct >> 5;
+ topQmf = topMdct >> 5;
+ }
+
+ if (band == ((int)numBands-1)) {
+ topQmf = (64);
+ }
+
+ for (bin = bottomQmf; bin < topQmf; bin++) {
+ FIXP_DBL drcFact1_mag = hDrcData->prevFact_mag[bin];
+ FIXP_DBL drcFact2_mag = fact_mag[band];
+
+ /* normalize scale factors */
+ if (hDrcData->prevFact_exp < maxShift) {
+ drcFact1_mag >>= maxShift - hDrcData->prevFact_exp;
+ }
+ if (fact_exp < maxShift) {
+ drcFact2_mag >>= maxShift - fact_exp;
+ }
+
+ /* interpolate */
+ drcFact_mag = fMult(alphaValue, drcFact2_mag) + fMult((FL2FXCONST_DBL(1.0f) - alphaValue), drcFact1_mag);
+
+ /* apply scaling */
+ qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag);
+ if (!useLP) {
+ qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag);
+ }
+
+ /* save previous factors */
+ if (col == (numQmfSubSamples>>1)-1) {
+ hDrcData->prevFact_mag[bin] = fact_mag[band];
+ }
+ }
+ }
+ else { /* short windows */
+ int startSample, stopSample;
+ FIXP_DBL invFrameSizeDiv8 = (frameLenFlag) ? (FIXP_DBL)0x1111111 : (FIXP_DBL)0x1000000;
+
+ if (frameLenFlag) {
+ /* 960 framing */
+ bottomMdct = 30/8 * (bottomMdct*8/30);
+ topMdct = 30/8 * (topMdct*8/30);
+ }
+ else {
+ /* 1024 framing */
+ bottomMdct &= ~0x03;
+ topMdct &= ~0x03;
+ }
+
+ /* startSample is truncated to the nearest corresponding start subsample in
+ the QMF of the short window bottom is present in:*/
+ startSample = ((fMultIfloor( invFrameSizeDiv8, bottomMdct ) & 0x7) * numQmfSubSamples) >> 3;
+
+ /* stopSample is rounded upwards to the nearest corresponding stop subsample
+ in the QMF of the short window top is present in. */
+ stopSample = ((fMultIceil( invFrameSizeDiv8, topMdct ) & 0xf) * numQmfSubSamples) >> 3;
+
+ bottomQmf = fMultIfloor( invFrameSizeDiv8, ((bottomMdct%(numQmfSubSamples<<2)) << 5) );
+ topQmf = fMultIfloor( invFrameSizeDiv8, ((topMdct%(numQmfSubSamples<<2)) << 5) );
+
+ /* extend last band */
+ if (band == ((int)numBands-1)) {
+ topQmf = (64);
+ stopSample = numQmfSubSamples;
+ }
+
+ if (topQmf == 0) {
+ topQmf = (64);
+ }
+
+ /* save previous factors */
+ if (stopSample == numQmfSubSamples) {
+ int tmpBottom = bottomQmf;
+
+ if (((numQmfSubSamples-1) & ~0x03) > startSample) {
+ tmpBottom = 0; /* band starts in previous short window */
+ }
+
+ for (bin = tmpBottom; bin < topQmf; bin++) {
+ hDrcData->prevFact_mag[bin] = fact_mag[band];
+ }
+ }
+
+ /* apply */
+ if ((col >= startSample) && (col < stopSample)) {
+ if ((col & ~0x03) > startSample) {
+ bottomQmf = 0; /* band starts in previous short window */
+ }
+ if (col < ((stopSample-1) & ~0x03)) {
+ topQmf = (64); /* band ends in next short window */
+ }
+
+ drcFact_mag = fact_mag[band];
+
+ /* normalize scale factor */
+ if (fact_exp < maxShift) {
+ drcFact_mag >>= maxShift - fact_exp;
+ }
+
+ /* apply scaling */
+ for (bin = bottomQmf; bin < topQmf; bin++) {
+ qmfRealSlot[bin] = fMult(qmfRealSlot[bin], drcFact_mag);
+ if (!useLP) {
+ qmfImagSlot[bin] = fMult(qmfImagSlot[bin], drcFact_mag);
+ }
+ }
+ }
+ }
+
+ bottomMdct = topMdct;
+ } /* end of bands loop */
+
+ if (col == (numQmfSubSamples>>1)-1) {
+ hDrcData->prevFact_exp = fact_exp;
+ }
+}
+
+
+/*!
+ \brief Apply DRC factors frame based.
+
+ \hDrcData Handle to DRC channel data.
+ \qmfRealSlot Pointer to real valued QMF data of the whole frame.
+ \qmfImagSlot Pointer to the imaginary QMF data of the whole frame.
+ \numQmfSubSamples Total number of time slots for one frame.
+ \scaleFactor Pointer to the out scale factor of the frame.
+
+ \return None.
+*/
+void sbrDecoder_drcApply (
+ HANDLE_SBR_DRC_CHANNEL hDrcData,
+ FIXP_DBL **QmfBufferReal,
+ FIXP_DBL **QmfBufferImag,
+ int numQmfSubSamples,
+ int *scaleFactor
+ )
+{
+ int col;
+ int maxShift = 0;
+
+ /* get max scale factor */
+ if (hDrcData->prevFact_exp > maxShift) {
+ maxShift = hDrcData->prevFact_exp;
+ }
+ if (hDrcData->currFact_exp > maxShift) {
+ maxShift = hDrcData->currFact_exp;
+ }
+ if (hDrcData->nextFact_exp > maxShift) {
+ maxShift = hDrcData->nextFact_exp;
+ }
+
+ for (col = 0; col < numQmfSubSamples; col++)
+ {
+ FIXP_DBL *qmfSlotReal = QmfBufferReal[col];
+ FIXP_DBL *qmfSlotImag = (QmfBufferImag == NULL) ? NULL : QmfBufferImag[col];
+
+ sbrDecoder_drcApplySlot (
+ hDrcData,
+ qmfSlotReal,
+ qmfSlotImag,
+ col,
+ numQmfSubSamples,
+ maxShift
+ );
+ }
+
+ *scaleFactor += maxShift;
+}
+
diff --git a/libSBRdec/src/sbrdec_drc.h b/libSBRdec/src/sbrdec_drc.h
new file mode 100644
index 0000000..2577e89
--- /dev/null
+++ b/libSBRdec/src/sbrdec_drc.h
@@ -0,0 +1,150 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************** MPEG-4 AAC Decoder **************************
+
+ Author(s): Christian Griebel
+ Description: Dynamic range control (DRC) decoder tool for SBR
+
+******************************************************************************/
+
+#ifndef _SBRDEC_DRC_H_
+#define _SBRDEC_DRC_H_
+
+#include "sbrdecoder.h"
+
+
+#define SBRDEC_MAX_DRC_CHANNELS (6)
+#define SBRDEC_MAX_DRC_BANDS ( 16 )
+
+typedef struct
+{
+ FIXP_DBL prevFact_mag[(64)];
+ INT prevFact_exp;
+
+ FIXP_DBL currFact_mag[SBRDEC_MAX_DRC_BANDS];
+ FIXP_DBL nextFact_mag[SBRDEC_MAX_DRC_BANDS];
+ INT currFact_exp;
+ INT nextFact_exp;
+
+ UINT numBandsCurr;
+ UINT numBandsNext;
+ USHORT bandTopCurr[SBRDEC_MAX_DRC_BANDS];
+ USHORT bandTopNext[SBRDEC_MAX_DRC_BANDS];
+
+ SHORT drcInterpolationSchemeCurr;
+ SHORT drcInterpolationSchemeNext;
+
+ SHORT enable;
+
+ UCHAR winSequenceCurr;
+ UCHAR winSequenceNext;
+
+} SBRDEC_DRC_CHANNEL;
+
+typedef SBRDEC_DRC_CHANNEL * HANDLE_SBR_DRC_CHANNEL;
+
+
+void sbrDecoder_drcInitChannel (
+ HANDLE_SBR_DRC_CHANNEL hDrcData );
+
+void sbrDecoder_drcUpdateChannel (
+ HANDLE_SBR_DRC_CHANNEL hDrcData );
+
+void sbrDecoder_drcApplySlot (
+ HANDLE_SBR_DRC_CHANNEL hDrcData,
+ FIXP_DBL *qmfRealSlot,
+ FIXP_DBL *qmfImagSlot,
+ int col,
+ int numQmfSubSamples,
+ int maxShift );
+
+void sbrDecoder_drcApply (
+ HANDLE_SBR_DRC_CHANNEL hDrcData,
+ FIXP_DBL **QmfBufferReal,
+ FIXP_DBL **QmfBufferImag,
+ int numQmfSubSamples,
+ int *scaleFactor );
+
+
+#endif /* _SBRDEC_DRC_H_ */
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
new file mode 100644
index 0000000..b877545
--- /dev/null
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -0,0 +1,805 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Frequency scale calculation
+*/
+
+#include "sbrdec_freq_sca.h"
+
+#include "transcendent.h"
+#include "sbr_rom.h"
+#include "env_extr.h"
+
+#include "genericStds.h" /* need log() for debug-code only */
+
+#define MAX_OCTAVE 29
+#define MAX_SECOND_REGION 50
+
+
+static int numberOfBands(FIXP_SGL bpo_div16, int start, int stop, int warpFlag);
+static void CalcBands(UCHAR * diff, UCHAR start, UCHAR stop, UCHAR num_bands);
+static SBR_ERROR modifyBands(UCHAR max_band, UCHAR * diff, UCHAR length);
+static void cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress);
+
+
+
+/*!
+ \brief Retrieve QMF-band where the SBR range starts
+
+ Convert startFreq which was read from the bitstream into a
+ QMF-channel number.
+
+ \return Number of start band
+*/
+static UCHAR
+getStartBand(UINT fs, /*!< Output sampling frequency */
+ UCHAR startFreq, /*!< Index to table of possible start bands */
+ UINT headerDataFlags) /*!< Info to SBR mode */
+{
+ INT band;
+ UINT fsMapped;
+
+ fsMapped = fs;
+
+ switch (fsMapped) {
+ case 48000:
+ band = FDK_sbrDecoder_sbr_start_freq_48[startFreq];
+ break;
+ case 44100:
+ band = FDK_sbrDecoder_sbr_start_freq_44[startFreq];
+ break;
+ case 32000:
+ band = FDK_sbrDecoder_sbr_start_freq_32[startFreq];
+ break;
+ case 24000:
+ band = FDK_sbrDecoder_sbr_start_freq_24[startFreq];
+ break;
+ case 22050:
+ band = FDK_sbrDecoder_sbr_start_freq_22[startFreq];
+ break;
+ case 16000:
+ band = FDK_sbrDecoder_sbr_start_freq_16[startFreq];
+ break;
+ default:
+ band = 255;
+ }
+
+ return band;
+}
+
+
+/*!
+ \brief Retrieve QMF-band where the SBR range starts
+
+ Convert startFreq which was read from the bitstream into a
+ QMF-channel number.
+
+ \return Number of start band
+*/
+static UCHAR
+getStopBand(UINT fs, /*!< Output sampling frequency */
+ UCHAR stopFreq, /*!< Index to table of possible start bands */
+ UINT headerDataFlags, /*!< Info to SBR mode */
+ UCHAR k0) /*!< Start freq index */
+{
+ UCHAR k2;
+
+ if (stopFreq < 14) {
+ INT stopMin;
+ UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
+ UCHAR *diff0 = diff_tot;
+ UCHAR *diff1 = diff_tot+MAX_OCTAVE;
+
+ if (fs < 32000) {
+ stopMin = (((2*6000*2*(64)) / fs) + 1) >> 1;
+ }
+ else {
+ if (fs < 64000) {
+ stopMin = (((2*8000*2*(64)) / fs) + 1) >> 1;
+ }
+ else {
+ stopMin = (((2*10000*2*(64)) / fs) + 1) >> 1;
+ }
+ }
+
+ /*
+ Choose a stop band between k1 and 64 depending on stopFreq (0..13),
+ based on a logarithmic scale.
+ The vectors diff0 and diff1 are used temporarily here.
+ */
+ CalcBands( diff0, stopMin, 64, 13);
+ shellsort( diff0, 13);
+ cumSum(stopMin, diff0, 13, diff1);
+ k2 = diff1[stopFreq];
+ }
+ else if (stopFreq==14)
+ k2 = 2*k0;
+ else
+ k2 = 3*k0;
+
+ /* Limit to Nyquist */
+ if (k2 > (64))
+ k2 = (64);
+
+
+ /* Range checks */
+ /* 1 <= difference <= 48; 1 <= fs <= 96000 */
+ if ( ((k2 - k0) > MAX_FREQ_COEFFS) || (k2 <= k0) ) {
+ return 255;
+ }
+
+ if (headerDataFlags & (SBRDEC_SYNTAX_USAC|SBRDEC_SYNTAX_RSVD50)) {
+ /* 1 <= difference <= 35; 42000 <= fs <= 96000 */
+ if ( (fs >= 42000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
+ return 255;
+ }
+ /* 1 <= difference <= 32; 46009 <= fs <= 96000 */
+ if ( (fs >= 46009) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
+ return 255;
+ }
+ }
+ else {
+ /* 1 <= difference <= 35; fs == 44100 */
+ if ( (fs == 44100) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS44100 ) ) {
+ return 255;
+ }
+ /* 1 <= difference <= 32; 48000 <= fs <= 96000 */
+ if ( (fs >= 48000) && ( (k2 - k0) > MAX_FREQ_COEFFS_FS48000 ) ) {
+ return 255;
+ }
+ }
+
+ return k2;
+}
+
+
+/*!
+ \brief Generates master frequency tables
+
+ Frequency tables are calculated according to the selected domain
+ (linear/logarithmic) and granularity.
+ IEC 14496-3 4.6.18.3.2.1
+
+ \return errorCode, 0 if successful
+*/
+SBR_ERROR
+sbrdecUpdateFreqScale(UCHAR * v_k_master, /*!< Master table to be created */
+ UCHAR *numMaster, /*!< Number of entries in master table */
+ UINT fs, /*!< SBR working sampling rate */
+ HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Control data from bitstream */
+ UINT flags)
+{
+ FIXP_SGL bpo_div16; /* bands_per_octave divided by 16 */
+ INT dk=0;
+
+ /* Internal variables */
+ UCHAR k0, k2, i;
+ UCHAR num_bands0 = 0;
+ UCHAR num_bands1 = 0;
+ UCHAR diff_tot[MAX_OCTAVE + MAX_SECOND_REGION];
+ UCHAR *diff0 = diff_tot;
+ UCHAR *diff1 = diff_tot+MAX_OCTAVE;
+ INT k2_achived;
+ INT k2_diff;
+ INT incr=0;
+
+ /*
+ Determine start band
+ */
+ k0 = getStartBand(fs, hHeaderData->bs_data.startFreq, flags);
+ if (k0 == 255) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ /*
+ Determine stop band
+ */
+ k2 = getStopBand(fs, hHeaderData->bs_data.stopFreq, flags, k0);
+ if (k2 == 255) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ if(hHeaderData->bs_data.freqScale>0) { /* Bark */
+ INT k1;
+
+ if(hHeaderData->bs_data.freqScale==1) {
+ bpo_div16 = FL2FXCONST_SGL(12.0f/16.0f);
+ }
+ else if(hHeaderData->bs_data.freqScale==2) {
+ bpo_div16 = FL2FXCONST_SGL(10.0f/16.0f);
+ }
+ else {
+ bpo_div16 = FL2FXCONST_SGL(8.0f/16.0f);
+ }
+
+
+ if( 1000 * k2 > 2245 * k0 ) { /* Two or more regions */
+ k1 = 2*k0;
+
+ num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
+ num_bands1 = numberOfBands(bpo_div16, k1, k2, hHeaderData->bs_data.alterScale );
+ if ( num_bands0 < 1) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+ if ( num_bands1 < 1 ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ CalcBands(diff0, k0, k1, num_bands0);
+ shellsort( diff0, num_bands0);
+ if (diff0[0] == 0) {
+#ifdef DEBUG_TOOLS
+#endif
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ cumSum(k0, diff0, num_bands0, v_k_master);
+
+ CalcBands(diff1, k1, k2, num_bands1);
+ shellsort( diff1, num_bands1);
+ if(diff0[num_bands0-1] > diff1[0]) {
+ SBR_ERROR err;
+
+ err = modifyBands(diff0[num_bands0-1],diff1, num_bands1);
+ if (err)
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ /* Add 2nd region */
+ cumSum(k1, diff1, num_bands1, &v_k_master[num_bands0]);
+ *numMaster = num_bands0 + num_bands1; /* Output nr of bands */
+
+ }
+ else { /* Only one region */
+ k1=k2;
+
+ num_bands0 = numberOfBands(bpo_div16, k0, k1, 0);
+ if ( num_bands0 < 1) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+ CalcBands(diff0, k0, k1, num_bands0);
+ shellsort(diff0, num_bands0);
+ if (diff0[0] == 0) {
+#ifdef DEBUG_TOOLS
+#endif
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ cumSum(k0, diff0, num_bands0, v_k_master);
+ *numMaster = num_bands0; /* Output nr of bands */
+
+ }
+ }
+ else { /* Linear mode */
+ if (hHeaderData->bs_data.alterScale==0) {
+ dk = 1;
+ /* FLOOR to get to few number of bands (next lower even number) */
+ num_bands0 = (k2 - k0) & 254;
+ } else {
+ dk = 2;
+ num_bands0 = ( ((k2 - k0) >> 1) + 1 ) & 254; /* ROUND to the closest fit */
+ }
+
+ if (num_bands0 < 1) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ /* We must return already here because 'i' can become negative below. */
+ }
+
+ k2_achived = k0 + num_bands0*dk;
+ k2_diff = k2 - k2_achived;
+
+ for(i=0;i<num_bands0;i++)
+ diff_tot[i] = dk;
+
+ /* If linear scale wasn't achieved */
+ /* and we got too wide SBR area */
+ if (k2_diff < 0) {
+ incr = 1;
+ i = 0;
+ }
+
+ /* If linear scale wasn't achieved */
+ /* and we got too small SBR area */
+ if (k2_diff > 0) {
+ incr = -1;
+ i = num_bands0-1;
+ }
+
+ /* Adjust diff vector to get sepc. SBR range */
+ while (k2_diff != 0) {
+ diff_tot[i] = diff_tot[i] - incr;
+ i = i + incr;
+ k2_diff = k2_diff + incr;
+ }
+
+ cumSum(k0, diff_tot, num_bands0, v_k_master);/* cumsum */
+ *numMaster = num_bands0; /* Output nr of bands */
+ }
+
+ if (*numMaster < 1) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+
+ /*
+ Print out the calculated table
+ */
+
+ return SBRDEC_OK;
+}
+
+
+/*!
+ \brief Calculate frequency ratio of one SBR band
+
+ All SBR bands should span a constant frequency range in the logarithmic
+ domain. This function calculates the ratio of any SBR band's upper and lower
+ frequency.
+
+ \return num_band-th root of k_start/k_stop
+*/
+static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands)
+{
+/* Scaled bandfactor and step 1 bit right to avoid overflow
+ * use double data type */
+ FIXP_DBL bandfactor = FL2FXCONST_DBL(0.25f); /* Start value */
+ FIXP_DBL step = FL2FXCONST_DBL(0.125f); /* Initial increment for factor */
+
+ int direction = 1;
+
+/* Because saturation can't be done in INT IIS,
+ * changed start and stop data type from FIXP_SGL to FIXP_DBL */
+ FIXP_DBL start = k_start << (DFRACT_BITS-8);
+ FIXP_DBL stop = k_stop << (DFRACT_BITS-8);
+
+ FIXP_DBL temp;
+
+ int j, i=0;
+
+ while ( step > FL2FXCONST_DBL(0.0f)) {
+ i++;
+ temp = stop;
+
+ /* Calculate temp^num_bands: */
+ for (j=0; j<num_bands; j++)
+ //temp = fMult(temp,bandfactor);
+ temp = fMultDiv2(temp,bandfactor)<<2;
+
+ if (temp<start) { /* Factor too strong, make it weaker */
+ if (direction == 0)
+ /* Halfen step. Right shift is not done as fract because otherwise the
+ lowest bit cannot be cleared due to rounding */
+ step = (FIXP_DBL)((LONG)step >> 1);
+ direction = 1;
+ bandfactor = bandfactor + step;
+ }
+ else { /* Factor is too weak: make it stronger */
+ if (direction == 1)
+ step = (FIXP_DBL)((LONG)step >> 1);
+ direction = 0;
+ bandfactor = bandfactor - step;
+ }
+
+ if (i>100) {
+ step = FL2FXCONST_DBL(0.0f);
+ }
+ }
+ return FX_DBL2FX_SGL(bandfactor<<1);
+}
+
+
+/*!
+ \brief Calculate number of SBR bands between start and stop band
+
+ Given the number of bands per octave, this function calculates how many
+ bands fit in the given frequency range.
+ When the warpFlag is set, the 'band density' is decreased by a factor
+ of 1/1.3
+
+ \return number of bands
+*/
+static int
+numberOfBands(FIXP_SGL bpo_div16, /*!< Input: number of bands per octave divided by 16 */
+ int start, /*!< First QMF band of SBR frequency range */
+ int stop, /*!< Last QMF band of SBR frequency range + 1 */
+ int warpFlag) /*!< Stretching flag */
+{
+ FIXP_SGL num_bands_div128;
+ int num_bands;
+
+ num_bands_div128 = FX_DBL2FX_SGL(fMult(FDK_getNumOctavesDiv8(start,stop),bpo_div16));
+
+ if (warpFlag) {
+ /* Apply the warp factor of 1.3 to get wider bands. We use a value
+ of 32768/25200 instead of the exact value to avoid critical cases
+ of rounding.
+ */
+ num_bands_div128 = FX_DBL2FX_SGL(fMult(num_bands_div128, FL2FXCONST_SGL(25200.0/32768.0)));
+ }
+
+ /* add scaled 1 for rounding to even numbers: */
+ num_bands_div128 = num_bands_div128 + FL2FXCONST_SGL( 1.0f/128.0f );
+ /* scale back to right aligned integer and double the value: */
+ num_bands = 2 * ((LONG)num_bands_div128 >> (FRACT_BITS - 7));
+
+ return(num_bands);
+}
+
+
+/*!
+ \brief Calculate width of SBR bands
+
+ Given the desired number of bands within the SBR frequency range,
+ this function calculates the width of each SBR band in QMF channels.
+ The bands get wider from start to stop (bark scale).
+*/
+static void
+CalcBands(UCHAR * diff, /*!< Vector of widths to be calculated */
+ UCHAR start, /*!< Lower end of subband range */
+ UCHAR stop, /*!< Upper end of subband range */
+ UCHAR num_bands) /*!< Desired number of bands */
+{
+ int i;
+ int previous;
+ int current;
+ FIXP_SGL exact, temp;
+ FIXP_SGL bandfactor = calcFactorPerBand(start, stop, num_bands);
+
+ previous = stop; /* Start with highest QMF channel */
+ exact = (FIXP_SGL)(stop << (FRACT_BITS-8)); /* Shift left to gain some accuracy */
+
+ for(i=num_bands-1; i>=0; i--) {
+ /* Calculate border of next lower sbr band */
+ exact = FX_DBL2FX_SGL(fMult(exact,bandfactor));
+
+ /* Add scaled 0.5 for rounding:
+ We use a value 128/256 instead of 0.5 to avoid some critical cases of rounding. */
+ temp = exact + FL2FXCONST_SGL(128.0/32768.0);
+
+ /* scale back to right alinged integer: */
+ current = (LONG)temp >> (FRACT_BITS-8);
+
+ /* Save width of band i */
+ diff[i] = previous - current;
+ previous = current;
+ }
+}
+
+
+/*!
+ \brief Calculate cumulated sum vector from delta vector
+*/
+static void
+cumSum(UCHAR start_value, UCHAR* diff, UCHAR length, UCHAR *start_adress)
+{
+ int i;
+ start_adress[0]=start_value;
+ for(i=1; i<=length; i++)
+ start_adress[i] = start_adress[i-1] + diff[i-1];
+}
+
+
+/*!
+ \brief Adapt width of frequency bands in the second region
+
+ If SBR spans more than 2 octaves, the upper part of a bark-frequency-scale
+ is calculated separately. This function tries to avoid that the second region
+ starts with a band smaller than the highest band of the first region.
+*/
+static SBR_ERROR
+modifyBands(UCHAR max_band_previous, UCHAR * diff, UCHAR length)
+{
+ int change = max_band_previous - diff[0];
+
+ /* Limit the change so that the last band cannot get narrower than the first one */
+ if ( change > (diff[length-1]-diff[0])>>1 )
+ change = (diff[length-1]-diff[0])>>1;
+
+ diff[0] += change;
+ diff[length-1] -= change;
+ shellsort(diff, length);
+
+ return SBRDEC_OK;
+}
+
+
+/*!
+ \brief Update high resolution frequency band table
+*/
+static void
+sbrdecUpdateHiRes(UCHAR * h_hires,
+ UCHAR * num_hires,
+ UCHAR * v_k_master,
+ UCHAR num_bands,
+ UCHAR xover_band)
+{
+ UCHAR i;
+
+ *num_hires = num_bands-xover_band;
+
+ for(i=xover_band; i<=num_bands; i++) {
+ h_hires[i-xover_band] = v_k_master[i];
+ }
+}
+
+
+/*!
+ \brief Build low resolution table out of high resolution table
+*/
+static void
+sbrdecUpdateLoRes(UCHAR * h_lores,
+ UCHAR * num_lores,
+ UCHAR * h_hires,
+ UCHAR num_hires)
+{
+ UCHAR i;
+
+ if( (num_hires & 1) == 0) {
+ /* If even number of hires bands */
+ *num_lores = num_hires >> 1;
+ /* Use every second lores=hires[0,2,4...] */
+ for(i=0; i<=*num_lores; i++)
+ h_lores[i] = h_hires[i*2];
+ }
+ else {
+ /* Odd number of hires, which means xover is odd */
+ *num_lores = (num_hires+1) >> 1;
+ /* Use lores=hires[0,1,3,5 ...] */
+ h_lores[0] = h_hires[0];
+ for(i=1; i<=*num_lores; i++) {
+ h_lores[i] = h_hires[i*2-1];
+ }
+ }
+}
+
+
+/*!
+ \brief Derive a low-resolution frequency-table from the master frequency table
+*/
+void
+sbrdecDownSampleLoRes(UCHAR *v_result,
+ UCHAR num_result,
+ UCHAR *freqBandTableRef,
+ UCHAR num_Ref)
+{
+ int step;
+ int i,j;
+ int org_length,result_length;
+ int v_index[MAX_FREQ_COEFFS>>1];
+
+ /* init */
+ org_length = num_Ref;
+ result_length = num_result;
+
+ v_index[0] = 0; /* Always use left border */
+ i=0;
+ while(org_length > 0) {
+ /* Create downsample vector */
+ i++;
+ step = org_length / result_length;
+ org_length = org_length - step;
+ result_length--;
+ v_index[i] = v_index[i-1] + step;
+ }
+
+ for(j=0;j<=i;j++) {
+ /* Use downsample vector to index LoResolution vector */
+ v_result[j]=freqBandTableRef[v_index[j]];
+ }
+
+}
+
+
+/*!
+ \brief Sorting routine
+*/
+void shellsort(UCHAR *in, UCHAR n)
+{
+
+ int i, j, v, w;
+ int inc = 1;
+
+ do
+ inc = 3 * inc + 1;
+ while (inc <= n);
+
+ do {
+ inc = inc / 3;
+ for (i = inc; i < n; i++) {
+ v = in[i];
+ j = i;
+ while ((w=in[j-inc]) > v) {
+ in[j] = w;
+ j -= inc;
+ if (j < inc)
+ break;
+ }
+ in[j] = v;
+ }
+ } while (inc > 1);
+
+}
+
+
+
+/*!
+ \brief Reset frequency band tables
+ \return errorCode, 0 if successful
+*/
+SBR_ERROR
+resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags)
+{
+ SBR_ERROR err = SBRDEC_OK;
+ int k2,kx, lsb, usb;
+ int intTemp;
+ UCHAR nBandsLo, nBandsHi;
+ HANDLE_FREQ_BAND_DATA hFreq = &hHeaderData->freqBandData;
+
+ /* Calculate master frequency function */
+ err = sbrdecUpdateFreqScale(hFreq->v_k_master,
+ &hFreq->numMaster,
+ hHeaderData->sbrProcSmplRate,
+ hHeaderData,
+ flags);
+
+ if ( err || (hHeaderData->bs_info.xover_band > hFreq->numMaster) ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ /* Derive Hiresolution from master frequency function */
+ sbrdecUpdateHiRes(hFreq->freqBandTable[1], &nBandsHi, hFreq->v_k_master, hFreq->numMaster, hHeaderData->bs_info.xover_band );
+ /* Derive Loresolution from Hiresolution */
+ sbrdecUpdateLoRes(hFreq->freqBandTable[0], &nBandsLo, hFreq->freqBandTable[1], nBandsHi);
+
+
+ hFreq->nSfb[0] = nBandsLo;
+ hFreq->nSfb[1] = nBandsHi;
+
+ /* Check index to freqBandTable[0] */
+ if ( !(nBandsLo > 0) || (nBandsLo > (MAX_FREQ_COEFFS>>1)) ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ lsb = hFreq->freqBandTable[0][0];
+ usb = hFreq->freqBandTable[0][nBandsLo];
+
+ /* Additional check for lsb */
+ if ( (lsb > (32)) || (lsb >= usb) ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+
+ /* Calculate number of noise bands */
+
+ k2 = hFreq->freqBandTable[1][nBandsHi];
+ kx = hFreq->freqBandTable[1][0];
+
+ if (hHeaderData->bs_data.noise_bands == 0)
+ {
+ hFreq->nNfb = 1;
+ }
+ else /* Calculate no of noise bands 1,2 or 3 bands/octave */
+ {
+ /* Fetch number of octaves divided by 32 */
+ intTemp = (LONG)FDK_getNumOctavesDiv8(kx,k2) >> 2;
+
+ /* Integer-Multiplication with number of bands: */
+ intTemp = intTemp * hHeaderData->bs_data.noise_bands;
+
+ /* Add scaled 0.5 for rounding: */
+ intTemp = intTemp + (LONG)FL2FXCONST_SGL(0.5f/32.0f);
+
+ /* Convert to right-aligned integer: */
+ intTemp = intTemp >> (FRACT_BITS - 1 /*sign*/ - 5 /* rescale */);
+
+ /* Compare with float calculation */
+ FDK_ASSERT( intTemp == (int)((hHeaderData->bs_data.noise_bands * FDKlog( (float)k2/kx) / (float)(FDKlog(2.0)))+0.5) );
+
+ if( intTemp==0)
+ intTemp=1;
+
+ hFreq->nNfb = intTemp;
+ }
+
+ hFreq->nInvfBands = hFreq->nNfb;
+
+ if( hFreq->nNfb > MAX_NOISE_COEFFS ) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ /* Get noise bands */
+ sbrdecDownSampleLoRes(hFreq->freqBandTableNoise,
+ hFreq->nNfb,
+ hFreq->freqBandTable[0],
+ nBandsLo);
+
+
+
+
+ hFreq->lowSubband = lsb;
+ hFreq->highSubband = usb;
+
+ return SBRDEC_OK;
+}
diff --git a/libSBRdec/src/sbrdec_freq_sca.h b/libSBRdec/src/sbrdec_freq_sca.h
new file mode 100644
index 0000000..eebdd52
--- /dev/null
+++ b/libSBRdec/src/sbrdec_freq_sca.h
@@ -0,0 +1,107 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief Frequency scale prototypes
+*/
+#ifndef __FREQ_SCA_H
+#define __FREQ_SCA_H
+
+#include "sbrdecoder.h"
+#include "env_extr.h"
+
+int
+sbrdecUpdateFreqScale(UCHAR * v_k_master,
+ UCHAR *numMaster,
+ HANDLE_SBR_HEADER_DATA headerData);
+
+void sbrdecDownSampleLoRes(UCHAR *v_result, UCHAR num_result,
+ UCHAR *freqBandTableRef, UCHAR num_Ref);
+
+void shellsort(UCHAR *in, UCHAR n);
+
+SBR_ERROR
+resetFreqBandTables(HANDLE_SBR_HEADER_DATA hHeaderData, const UINT flags);
+
+#endif
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
new file mode 100644
index 0000000..a40e5ba
--- /dev/null
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -0,0 +1,1527 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief SBR decoder frontend
+ This module provides a frontend to the SBR decoder. The function openSBR() is called for
+ initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the
+ required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()],
+ and to finally apply SBR to the current frame [sbr_dec()].
+
+ \sa sbrDecoder_Apply(), \ref documentationOverview
+*/
+
+/*!
+ \page documentationOverview Overview of important information resources and source code documentation
+
+ The primary source code documentation is based on generated and cross-referenced HTML files using
+ <a HREF="http://www.doxygen.org">doxygen</a>. As part of this documentation
+ you can find more extensive descriptions about key concepts and algorithms at the following locations:
+
+ <h2>Programming</h2>
+
+ \li Buffer management: sbrDecoder_Apply() and sbr_dec()
+ \li Internal scale factors to maximize SNR on fixed point processors: #QMF_SCALE_FACTOR
+ \li Special mantissa-exponent format: Created in requantizeEnvelopeData() and used in calculateSbrEnvelope()
+
+ <h2>Algorithmic details</h2>
+ \li About the SBR data format: \ref SBR_HEADER_ELEMENT and \ref SBR_STANDARD_ELEMENT
+ \li Details about the bitstream decoder: env_extr.cpp
+ \li Details about the QMF filterbank and the provided polyphase implementation: qmf_dec.cpp
+ \li Details about the transposer: lpp_tran.cpp
+ \li Details about the envelope adjuster: env_calc.cpp
+
+*/
+
+#include "sbrdecoder.h"
+
+#include "FDK_bitstream.h"
+
+#include "sbrdec_freq_sca.h"
+#include "env_extr.h"
+#include "sbr_dec.h"
+#include "env_dec.h"
+#include "sbr_crc.h"
+#include "sbr_ram.h"
+#include "sbr_rom.h"
+#include "lpp_tran.h"
+#include "transcendent.h"
+
+
+#include "sbrdec_drc.h"
+
+#include "psbitdec.h"
+
+
+/* Decoder library info */
+#define SBRDECODER_LIB_VL0 2
+#define SBRDECODER_LIB_VL1 1
+#define SBRDECODER_LIB_VL2 2
+#define SBRDECODER_LIB_TITLE "SBR Decoder"
+#define SBRDECODER_LIB_BUILD_DATE __DATE__
+#define SBRDECODER_LIB_BUILD_TIME __TIME__
+
+
+
+
+static UCHAR getHeaderSlot( UCHAR currentSlot, UCHAR hdrSlotUsage[(1)+1] )
+{
+ UINT occupied = 0;
+ int s;
+ UCHAR slot = hdrSlotUsage[currentSlot];
+
+ FDK_ASSERT((1)+1 < 32);
+
+ for (s = 0; s < (1)+1; s++) {
+ if ( (hdrSlotUsage[s] == slot)
+ && (s != slot) ) {
+ occupied = 1;
+ break;
+ }
+ }
+
+ if (occupied) {
+ occupied = 0;
+
+ for (s = 0; s < (1)+1; s++) {
+ occupied |= 1 << hdrSlotUsage[s];
+ }
+ for (s = 0; s < (1)+1; s++) {
+ if ( !(occupied & 0x1) ) {
+ slot = s;
+ break;
+ }
+ occupied >>= 1;
+ }
+ }
+
+ return slot;
+}
+
+static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_DATA hSrc )
+{
+ /* copy the whole header memory (including pointers) */
+ FDKmemcpy( hDst, hSrc, sizeof(SBR_HEADER_DATA) );
+
+ /* update pointers */
+ hDst->freqBandData.freqBandTable[0] = hDst->freqBandData.freqBandTableLo;
+ hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi;
+}
+
+
+/*!
+ \brief Reset SBR decoder.
+
+ Reset should only be called if SBR has been sucessfully detected by
+ an appropriate checkForPayload() function.
+
+ \return Error code.
+*/
+static
+SBR_ERROR sbrDecoder_ResetElement (
+ HANDLE_SBRDECODER self,
+ int sampleRateIn,
+ int sampleRateOut,
+ int samplesPerFrame,
+ const MP4_ELEMENT_ID elementID,
+ const int elementIndex,
+ const int overlap
+ )
+{
+ SBR_ERROR sbrError = SBRDEC_OK;
+ HANDLE_SBR_HEADER_DATA hSbrHeader;
+ UINT qmfFlags = 0;
+
+ int i, synDownsampleFac;
+
+ /* Check in/out samplerates */
+ if ( sampleRateIn < 6400
+ || sampleRateIn > 24000
+ )
+ {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+
+ if ( sampleRateOut > 48000 )
+ {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+
+ /* Set QMF mode flags */
+ if (self->flags & SBRDEC_LOW_POWER)
+ qmfFlags |= QMF_FLAG_LP;
+
+ if (self->coreCodec == AOT_ER_AAC_ELD) {
+ if (self->flags & SBRDEC_LD_MPS_QMF) {
+ qmfFlags |= QMF_FLAG_MPSLDFB;
+ } else {
+ qmfFlags |= QMF_FLAG_CLDFB;
+ }
+ }
+
+ /* Set downsampling factor for synthesis filter bank */
+ if (sampleRateOut == 0)
+ {
+ /* no single rate mode */
+ sampleRateOut = sampleRateIn<<1; /* In case of implicit signalling, assume dual rate SBR */
+ }
+
+ if ( sampleRateIn == sampleRateOut ) {
+ synDownsampleFac = 2;
+ } else {
+ synDownsampleFac = 1;
+ }
+
+ self->synDownsampleFac = synDownsampleFac;
+ self->sampleRateOut = sampleRateOut;
+
+ {
+ int i;
+
+ for (i = 0; i < (1)+1; i++)
+ {
+ hSbrHeader = &(self->sbrHeader[elementIndex][i]);
+
+ /* init a default header such that we can at least do upsampling later */
+ sbrError = initHeaderData(
+ hSbrHeader,
+ sampleRateIn,
+ sampleRateOut,
+ samplesPerFrame,
+ self->flags
+ );
+ }
+ }
+
+ if (sbrError != SBRDEC_OK) {
+ goto bail;
+ }
+
+ /* Init SBR channels going to be assigned to a SBR element */
+ {
+ int ch;
+
+ for (ch=0; ch<self->pSbrElement[elementIndex]->nChannels; ch++)
+ {
+ /* and create sbrDec */
+ sbrError = createSbrDec (self->pSbrElement[elementIndex]->pSbrChannel[ch],
+ hSbrHeader,
+ &self->pSbrElement[elementIndex]->transposerSettings,
+ synDownsampleFac,
+ qmfFlags,
+ self->flags,
+ overlap,
+ ch );
+
+ if (sbrError != SBRDEC_OK) {
+ goto bail;
+ }
+ }
+ }
+
+ //FDKmemclear(sbr_OverlapBuffer, sizeof(sbr_OverlapBuffer));
+
+ if (self->numSbrElements == 1) {
+ switch ( self->coreCodec ) {
+ case AOT_AAC_LC:
+ case AOT_SBR:
+ case AOT_PS:
+ case AOT_ER_AAC_SCAL:
+ case AOT_DRM_AAC:
+ case AOT_DRM_SURROUND:
+ if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
+ sbrError = SBRDEC_CREATE_ERROR;
+ goto bail;
+ }
+ break;
+ default:
+ break;
+ }
+ }
+
+ /* Init frame delay slot handling */
+ self->pSbrElement[elementIndex]->useFrameSlot = 0;
+ for (i = 0; i < ((1)+1); i++) {
+ self->pSbrElement[elementIndex]->useHeaderSlot[i] = i;
+ }
+
+bail:
+
+ return sbrError;
+}
+
+
+SBR_ERROR sbrDecoder_Open ( HANDLE_SBRDECODER * pSelf )
+{
+ HANDLE_SBRDECODER self = NULL;
+ SBR_ERROR sbrError = SBRDEC_OK;
+
+ /* Get memory for this instance */
+ self = GetRam_SbrDecoder();
+ if (self == NULL) {
+ sbrError = SBRDEC_MEM_ALLOC_FAILED;
+ goto bail;
+ }
+
+ self->workBuffer1 = GetRam_SbrDecWorkBuffer1();
+ self->workBuffer2 = GetRam_SbrDecWorkBuffer2();
+
+ if ( self->workBuffer1 == NULL
+ || self->workBuffer2 == NULL )
+ {
+ sbrError = SBRDEC_MEM_ALLOC_FAILED;
+ goto bail;
+ }
+
+ /*
+ Already zero because of calloc
+ self->numSbrElements = 0;
+ self->numSbrChannels = 0;
+ self->codecFrameSize = 0;
+ */
+
+ self->numDelayFrames = (1); /* set to the max value by default */
+
+ *pSelf = self;
+
+bail:
+ return sbrError;
+}
+
+/**
+ * \brief determine if the given core codec AOT can be processed or not.
+ * \param coreCodec core codec audio object type.
+ * \return 1 if SBR can be processed, 0 if SBR cannot be processed/applied.
+ */
+static
+int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec)
+{
+ switch (coreCodec) {
+ case AOT_AAC_LC:
+ case AOT_SBR:
+ case AOT_PS:
+ case AOT_ER_AAC_SCAL:
+ case AOT_ER_AAC_ELD:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static
+void sbrDecoder_DestroyElement (
+ HANDLE_SBRDECODER self,
+ const int elementIndex
+ )
+{
+ if (self->pSbrElement[elementIndex] != NULL) {
+ int ch;
+
+ for (ch=0; ch<SBRDEC_MAX_CH_PER_ELEMENT; ch++) {
+ if (self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) {
+ deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+ FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+ self->numSbrChannels -= 1;
+ }
+ }
+ FreeRam_SbrDecElement( &self->pSbrElement[elementIndex] );
+ self->numSbrElements -= 1;
+ }
+}
+
+
+SBR_ERROR sbrDecoder_InitElement (
+ HANDLE_SBRDECODER self,
+ const int sampleRateIn,
+ const int sampleRateOut,
+ const int samplesPerFrame,
+ const AUDIO_OBJECT_TYPE coreCodec,
+ const MP4_ELEMENT_ID elementID,
+ const int elementIndex
+ )
+{
+ SBR_ERROR sbrError = SBRDEC_OK;
+ int chCnt=0;
+ int nSbrElementsStart = self->numSbrElements;
+
+ /* Check core codec AOT */
+ if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (4)) {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+
+ if ( elementID != ID_SCE && elementID != ID_CPE && elementID != ID_LFE )
+ {
+ sbrError = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+
+ if ( self->sampleRateIn == sampleRateIn
+ && self->codecFrameSize == samplesPerFrame
+ && self->coreCodec == coreCodec
+ && self->pSbrElement[elementIndex] != NULL
+ && self->pSbrElement[elementIndex]->elementID == elementID
+ )
+ {
+ /* Nothing to do */
+ return SBRDEC_OK;
+ }
+
+ self->sampleRateIn = sampleRateIn;
+ self->codecFrameSize = samplesPerFrame;
+ self->coreCodec = coreCodec;
+
+ self->flags = 0;
+ self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0;
+
+ /* Init SBR elements */
+ {
+ int elChannels, ch;
+
+ if (self->pSbrElement[elementIndex] == NULL) {
+ self->pSbrElement[elementIndex] = GetRam_SbrDecElement(elementIndex);
+ if (self->pSbrElement[elementIndex] == NULL) {
+ sbrError = SBRDEC_MEM_ALLOC_FAILED;
+ goto bail;
+ }
+ self->numSbrElements ++;
+ } else {
+ self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels;
+ }
+
+ /* Save element ID for sanity checks and to have a fallback for concealment. */
+ self->pSbrElement[elementIndex]->elementID = elementID;
+
+ /* Determine amount of channels for this element */
+ switch (elementID) {
+ case ID_NONE:
+ case ID_CPE: elChannels=2;
+ break;
+ case ID_LFE:
+ case ID_SCE: elChannels=1;
+ break;
+ default: elChannels=0;
+ break;
+ }
+
+ /* Handle case of Parametric Stereo */
+ if ( elementIndex == 0 && elementID == ID_SCE ) {
+ switch (coreCodec) {
+ case AOT_AAC_LC:
+ case AOT_SBR:
+ case AOT_PS:
+ case AOT_ER_AAC_SCAL:
+ case AOT_DRM_AAC:
+ case AOT_DRM_SURROUND:
+ elChannels = 2;
+ break;
+ default:
+ break;
+ }
+ }
+
+ self->pSbrElement[elementIndex]->nChannels = elChannels;
+
+ for (ch=0; ch<elChannels; ch++)
+ {
+ if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) {
+ self->pSbrElement[elementIndex]->pSbrChannel[ch] = GetRam_SbrDecChannel(chCnt);
+ if (self->pSbrElement[elementIndex]->pSbrChannel[ch] == NULL) {
+ sbrError = SBRDEC_MEM_ALLOC_FAILED;
+ goto bail;
+ }
+ }
+ self->numSbrChannels ++;
+
+ sbrDecoder_drcInitChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.sbrDrcChannel );
+
+ /* Add reference pointer to workbuffers. */
+ self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer1 = self->workBuffer1;
+ self->pSbrElement[elementIndex]->pSbrChannel[ch]->SbrDec.WorkBuffer2 = self->workBuffer2;
+ chCnt++;
+ }
+ if (elChannels == 1 && self->pSbrElement[elementIndex]->pSbrChannel[ch] != NULL) {
+ deleteSbrDec( self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+ FreeRam_SbrDecChannel( &self->pSbrElement[elementIndex]->pSbrChannel[ch] );
+ }
+ }
+
+ /* clear error flags for all delay slots */
+ FDKmemclear(self->pSbrElement[elementIndex]->frameErrorFlag, ((1)+1)*sizeof(UCHAR));
+
+ /* Initialize this instance */
+ sbrError = sbrDecoder_ResetElement(
+ self,
+ sampleRateIn,
+ sampleRateOut,
+ samplesPerFrame,
+ elementID,
+ elementIndex,
+ (coreCodec == AOT_ER_AAC_ELD) ? 0 : (6)
+ );
+
+
+
+bail:
+ if (sbrError != SBRDEC_OK) {
+ if (nSbrElementsStart < self->numSbrElements) {
+ /* Free the memory allocated for this element */
+ sbrDecoder_DestroyElement( self, elementIndex );
+ } else if (self->pSbrElement[elementIndex] != NULL) {
+ /* Set error flag to trigger concealment */
+ self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;;
+ }
+ }
+
+ return sbrError;
+}
+
+/**
+ * \brief Apply decoded SBR header for one element.
+ * \param self SBR decoder instance handle
+ * \param hSbrHeader SBR header handle to be processed.
+ * \param hSbrChannel pointer array to the SBR element channels corresponding to the SBR header.
+ * \param headerStatus header status value returned from SBR header parser.
+ * \param numElementChannels amount of channels for the SBR element whos header is to be processed.
+ */
+static
+SBR_ERROR sbrDecoder_HeaderUpdate(
+ HANDLE_SBRDECODER self,
+ HANDLE_SBR_HEADER_DATA hSbrHeader,
+ SBR_HEADER_STATUS headerStatus,
+ HANDLE_SBR_CHANNEL hSbrChannel[],
+ const int numElementChannels
+ )
+{
+ SBR_ERROR errorStatus = SBRDEC_OK;
+
+ /*
+ change of control data, reset decoder
+ */
+ errorStatus = resetFreqBandTables(hSbrHeader, self->flags);
+
+ if (errorStatus == SBRDEC_OK) {
+ if (hSbrHeader->syncState == UPSAMPLING && headerStatus != HEADER_RESET)
+ {
+ /* As the default header would limit the frequency range,
+ lowSubband and highSubband must be patched. */
+ hSbrHeader->freqBandData.lowSubband = hSbrHeader->numberOfAnalysisBands;
+ hSbrHeader->freqBandData.highSubband = hSbrHeader->numberOfAnalysisBands;
+ }
+
+ /* Trigger a reset before processing this slot */
+ hSbrHeader->status |= SBRDEC_HDR_STAT_RESET;
+ }
+
+ return errorStatus;
+}
+
+INT sbrDecoder_Header (
+ HANDLE_SBRDECODER self,
+ HANDLE_FDK_BITSTREAM hBs,
+ const INT sampleRateIn,
+ const INT sampleRateOut,
+ const INT samplesPerFrame,
+ const AUDIO_OBJECT_TYPE coreCodec,
+ const MP4_ELEMENT_ID elementID,
+ const INT elementIndex
+ )
+{
+ SBR_HEADER_STATUS headerStatus;
+ HANDLE_SBR_HEADER_DATA hSbrHeader;
+ SBR_ERROR sbrError = SBRDEC_OK;
+ int headerIndex;
+
+ if ( self == NULL || elementIndex > (4) )
+ {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ if (! sbrDecoder_isCoreCodecValid(coreCodec)) {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+
+ sbrError = sbrDecoder_InitElement(
+ self,
+ sampleRateIn,
+ sampleRateOut,
+ samplesPerFrame,
+ coreCodec,
+ elementID,
+ elementIndex
+ );
+
+ if (sbrError != SBRDEC_OK) {
+ goto bail;
+ }
+
+ headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
+ self->pSbrElement[elementIndex]->useHeaderSlot);
+ hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]);
+
+ headerStatus = sbrGetHeaderData ( hSbrHeader,
+ hBs,
+ self->flags,
+ 0);
+
+
+ {
+ SBR_DECODER_ELEMENT *pSbrElement;
+
+ pSbrElement = self->pSbrElement[elementIndex];
+
+ /* Sanity check */
+ if (pSbrElement != NULL) {
+ if ( (elementID == ID_CPE && pSbrElement->nChannels != 2)
+ || (elementID != ID_CPE && pSbrElement->nChannels != 1) )
+ {
+ return SBRDEC_UNSUPPORTED_CONFIG;
+ }
+ if ( headerStatus == HEADER_RESET ) {
+
+ sbrError = sbrDecoder_HeaderUpdate(
+ self,
+ hSbrHeader,
+ headerStatus,
+ pSbrElement->pSbrChannel,
+ pSbrElement->nChannels
+ );
+
+ if (sbrError == SBRDEC_OK) {
+ hSbrHeader->syncState = SBR_HEADER;
+ hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ }
+ /* else {
+ Since we already have overwritten the old SBR header the only way out is UPSAMPLING!
+ This will be prepared in the next step.
+ } */
+ }
+ }
+ }
+bail:
+ return sbrError;
+}
+
+
+SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER self,
+ const SBRDEC_PARAM param,
+ const INT value )
+{
+ SBR_ERROR errorStatus = SBRDEC_OK;
+
+ /* configure the subsystems */
+ switch (param)
+ {
+ case SBR_SYSTEM_BITSTREAM_DELAY:
+ if (value < 0 || value > (1)) {
+ errorStatus = SBRDEC_SET_PARAM_FAIL;
+ break;
+ }
+ if (self == NULL) {
+ errorStatus = SBRDEC_NOT_INITIALIZED;
+ } else {
+ self->numDelayFrames = (UCHAR)value;
+ }
+ break;
+ case SBR_QMF_MODE:
+ if (self == NULL) {
+ errorStatus = SBRDEC_NOT_INITIALIZED;
+ } else {
+ if (value == 1) {
+ self->flags |= SBRDEC_LOW_POWER;
+ } else {
+ self->flags &= ~SBRDEC_LOW_POWER;
+ }
+ }
+ break;
+ case SBR_LD_QMF_TIME_ALIGN:
+ if (self == NULL) {
+ errorStatus = SBRDEC_NOT_INITIALIZED;
+ } else {
+ if (value == 1) {
+ self->flags |= SBRDEC_LD_MPS_QMF;
+ } else {
+ self->flags &= ~SBRDEC_LD_MPS_QMF;
+ }
+ }
+ break;
+ case SBR_BS_INTERRUPTION:
+ {
+ int elementIndex;
+ /* Loop over SBR elements */
+ for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++)
+ {
+ HANDLE_SBR_HEADER_DATA hSbrHeader;
+ int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
+ self->pSbrElement[elementIndex]->useHeaderSlot);
+
+ hSbrHeader = &(self->sbrHeader[elementIndex][headerIndex]);
+
+ /* Set sync state UPSAMPLING for the corresponding slot.
+ This switches off bitstream parsing until a new header arrives. */
+ hSbrHeader->syncState = UPSAMPLING;
+ hSbrHeader->status |= SBRDEC_HDR_STAT_UPDATE;
+ }
+ }
+ break;
+ default:
+ errorStatus = SBRDEC_SET_PARAM_FAIL;
+ break;
+ } /* switch(param) */
+
+ return (errorStatus);
+}
+
+static
+SBRDEC_DRC_CHANNEL * sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, const INT channel )
+{
+ SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+ int elementIndex, elChanIdx=0, numCh=0;
+
+ for (elementIndex = 0; (elementIndex < (4)) && (numCh <= channel); elementIndex++)
+ {
+ SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex];
+ int c, elChannels;
+
+ elChanIdx = 0;
+ if (pSbrElement == NULL) break;
+
+ /* Determine amount of channels for this element */
+ switch (pSbrElement->elementID) {
+ case ID_CPE: elChannels = 2;
+ break;
+ case ID_LFE:
+ case ID_SCE: elChannels = 1;
+ break;
+ case ID_NONE:
+ default: elChannels = 0;
+ break;
+ }
+
+ /* Limit with actual allocated element channels */
+ elChannels = FDKmin(elChannels, pSbrElement->nChannels);
+
+ for (c = 0; (c < elChannels) && (numCh <= channel); c++) {
+ if (pSbrElement->pSbrChannel[elChanIdx] != NULL) {
+ numCh++;
+ elChanIdx++;
+ }
+ }
+ }
+ elementIndex -= 1;
+ elChanIdx -= 1;
+
+ if (elChanIdx < 0 || elementIndex < 0) {
+ return NULL;
+ }
+
+ if ( self->pSbrElement[elementIndex] != NULL ) {
+ if ( self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx] != NULL )
+ {
+ pSbrDrcChannelData = &self->pSbrElement[elementIndex]->pSbrChannel[elChanIdx]->SbrDec.sbrDrcChannel;
+ }
+ }
+
+ return (pSbrDrcChannelData);
+}
+
+SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER self,
+ INT ch,
+ UINT numBands,
+ FIXP_DBL *pNextFact_mag,
+ INT nextFact_exp,
+ SHORT drcInterpolationScheme,
+ UCHAR winSequence,
+ USHORT *pBandTop )
+{
+ SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+
+ if (self == NULL) {
+ return SBRDEC_NOT_INITIALIZED;
+ }
+ if (ch > (6) || pNextFact_mag == NULL) {
+ return SBRDEC_SET_PARAM_FAIL;
+ }
+
+ /* Find the right SBR channel */
+ pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch );
+
+ if ( pSbrDrcChannelData != NULL ) {
+ int i;
+
+ pSbrDrcChannelData->enable = 1;
+ pSbrDrcChannelData->numBandsNext = numBands;
+
+ pSbrDrcChannelData->winSequenceNext = winSequence;
+ pSbrDrcChannelData->drcInterpolationSchemeNext = drcInterpolationScheme;
+ pSbrDrcChannelData->nextFact_exp = nextFact_exp;
+
+ for (i = 0; i < (int)numBands; i++) {
+ pSbrDrcChannelData->bandTopNext[i] = pBandTop[i];
+ pSbrDrcChannelData->nextFact_mag[i] = pNextFact_mag[i];
+ }
+ }
+
+ return SBRDEC_OK;
+}
+
+
+void sbrDecoder_drcDisable ( HANDLE_SBRDECODER self,
+ INT ch )
+{
+ SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
+
+ if ( (self == NULL)
+ || (ch > (6))
+ || (self->numSbrElements == 0)
+ || (self->numSbrChannels == 0) ) {
+ return;
+ }
+
+ /* Find the right SBR channel */
+ pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch );
+
+ if ( pSbrDrcChannelData != NULL ) {
+ pSbrDrcChannelData->enable = 0;
+ }
+}
+
+
+
+SBR_ERROR sbrDecoder_Parse(
+ HANDLE_SBRDECODER self,
+ HANDLE_FDK_BITSTREAM hBs,
+ int *count,
+ int bsPayLen,
+ int crcFlag,
+ MP4_ELEMENT_ID prevElement,
+ int elementIndex,
+ int fGlobalIndependencyFlag
+ )
+{
+ SBR_DECODER_ELEMENT *hSbrElement;
+ HANDLE_SBR_HEADER_DATA hSbrHeader;
+ HANDLE_SBR_CHANNEL *pSbrChannel;
+
+ SBR_FRAME_DATA *hFrameDataLeft;
+ SBR_FRAME_DATA *hFrameDataRight;
+
+ SBR_ERROR errorStatus = SBRDEC_OK;
+ SBR_SYNC_STATE initialSyncState;
+ SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
+
+ INT startPos;
+ INT CRCLen = 0;
+
+ int stereo;
+ int fDoDecodeSbrData = 1;
+
+ int lastSlot, lastHdrSlot = 0, thisHdrSlot;
+
+ /* Remember start position of SBR element */
+ startPos = FDKgetValidBits(hBs);
+
+ /* SBR sanity checks */
+ if ( self == NULL || self->pSbrElement[elementIndex] == NULL ) {
+ errorStatus = SBRDEC_NOT_INITIALIZED;
+ goto bail;
+ }
+
+ hSbrElement = self->pSbrElement[elementIndex];
+
+ lastSlot = (hSbrElement->useFrameSlot > 0) ? hSbrElement->useFrameSlot-1 : self->numDelayFrames;
+ lastHdrSlot = hSbrElement->useHeaderSlot[lastSlot];
+ thisHdrSlot = getHeaderSlot( hSbrElement->useFrameSlot, hSbrElement->useHeaderSlot ); /* Get a free header slot not used by frames not processed yet. */
+
+ /* Assign the free slot to store a new header if there is one. */
+ hSbrHeader = &self->sbrHeader[elementIndex][thisHdrSlot];
+
+ pSbrChannel = hSbrElement->pSbrChannel;
+ stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
+
+ hFrameDataLeft = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+ hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+
+ initialSyncState = hSbrHeader->syncState;
+
+ /* reset PS flag; will be set after PS was found */
+ self->flags &= ~SBRDEC_PS_DECODED;
+
+ if (hSbrHeader->status & SBRDEC_HDR_STAT_UPDATE) {
+ /* Got a new header from extern (e.g. from an ASC) */
+ headerStatus = HEADER_OK;
+ hSbrHeader->status &= ~SBRDEC_HDR_STAT_UPDATE;
+ }
+ else if (thisHdrSlot != lastHdrSlot) {
+ /* Copy the last header into this slot otherwise the
+ header compare will trigger more HEADER_RESETs than needed. */
+ copySbrHeader( hSbrHeader, &self->sbrHeader[elementIndex][lastHdrSlot] );
+ }
+
+ /*
+ Check if bit stream data is valid and matches the element context
+ */
+ if ( ((prevElement != ID_SCE) && (prevElement != ID_CPE)) || prevElement != hSbrElement->elementID) {
+ /* In case of LFE we also land here, since there is no LFE SBR element (do upsampling only) */
+ fDoDecodeSbrData = 0;
+ }
+
+ if (fDoDecodeSbrData)
+ {
+ if ((INT)FDKgetValidBits(hBs) <= 0) {
+ fDoDecodeSbrData = 0;
+ }
+ }
+
+ /*
+ SBR CRC-check
+ */
+ if (fDoDecodeSbrData)
+ {
+ if (crcFlag == 1) {
+ switch (self->coreCodec) {
+ case AOT_ER_AAC_ELD:
+ FDKpushFor (hBs, 10);
+ /* check sbrcrc later: we don't know the payload length now */
+ break;
+ default:
+ CRCLen = bsPayLen - 10; /* change: 0 => i */
+ if (CRCLen < 0) {
+ fDoDecodeSbrData = 0;
+ } else {
+ fDoDecodeSbrData = SbrCrcCheck (hBs, CRCLen);
+ }
+ break;
+ }
+ }
+ } /* if (fDoDecodeSbrData) */
+
+ /*
+ Read in the header data and issue a reset if change occured
+ */
+ if (fDoDecodeSbrData)
+ {
+ int sbrHeaderPresent;
+
+ {
+ sbrHeaderPresent = FDKreadBit(hBs);
+ }
+
+ if ( sbrHeaderPresent ) {
+ headerStatus = sbrGetHeaderData (hSbrHeader,
+ hBs,
+ self->flags,
+ 1);
+ }
+
+ if (headerStatus == HEADER_RESET)
+ {
+ errorStatus = sbrDecoder_HeaderUpdate(
+ self,
+ hSbrHeader,
+ headerStatus,
+ pSbrChannel,
+ hSbrElement->nChannels
+ );
+
+ if (errorStatus == SBRDEC_OK) {
+ hSbrHeader->syncState = SBR_HEADER;
+ } else {
+ hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+ }
+ }
+
+ if (errorStatus != SBRDEC_OK) {
+ fDoDecodeSbrData = 0;
+ }
+ } /* if (fDoDecodeSbrData) */
+
+ /*
+ Print debugging output only if state has changed
+ */
+
+ /* read frame data */
+ if ((hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) {
+ int sbrFrameOk;
+ /* read the SBR element data */
+ if (stereo) {
+ sbrFrameOk = sbrGetChannelPairElement(hSbrHeader,
+ hFrameDataLeft,
+ hFrameDataRight,
+ hBs,
+ self->flags,
+ self->pSbrElement[elementIndex]->transposerSettings.overlap);
+ }
+ else {
+ if (self->hParametricStereoDec != NULL) {
+ /* update slot index for PS bitstream parsing */
+ self->hParametricStereoDec->bsLastSlot = self->hParametricStereoDec->bsReadSlot;
+ self->hParametricStereoDec->bsReadSlot = hSbrElement->useFrameSlot;
+ }
+ sbrFrameOk = sbrGetSingleChannelElement(hSbrHeader,
+ hFrameDataLeft,
+ hBs,
+ self->hParametricStereoDec,
+ self->flags,
+ self->pSbrElement[elementIndex]->transposerSettings.overlap);
+ }
+ if (!sbrFrameOk) {
+ fDoDecodeSbrData = 0;
+ }
+ else {
+ INT valBits;
+
+ if (bsPayLen > 0) {
+ valBits = bsPayLen - ((INT)startPos - (INT)FDKgetValidBits(hBs));
+ } else {
+ valBits = (INT)FDKgetValidBits(hBs);
+ }
+
+ if ( crcFlag == 1 ) {
+ switch (self->coreCodec) {
+ case AOT_ER_AAC_ELD:
+ {
+ /* late crc check for eld */
+ INT payloadbits = (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos;
+ INT crcLen = payloadbits - 10;
+ FDKpushBack(hBs, payloadbits);
+ fDoDecodeSbrData = SbrCrcCheck (hBs, crcLen);
+ FDKpushFor(hBs, crcLen);
+ }
+ break;
+ default:
+ break;
+ }
+ }
+
+ /* sanity check of remaining bits */
+ if (valBits < 0) {
+ fDoDecodeSbrData = 0;
+ } else {
+ switch (self->coreCodec) {
+ case AOT_SBR:
+ case AOT_PS:
+ case AOT_AAC_LC:
+ {
+ /* This sanity check is only meaningful with General Audio bitstreams */
+ int alignBits = valBits & 0x7;
+
+ if (valBits > alignBits) {
+ fDoDecodeSbrData = 0;
+ }
+ }
+ break;
+ default:
+ /* No sanity check available */
+ break;
+ }
+ }
+ }
+ }
+
+ if (!fDoDecodeSbrData) {
+ /* Set error flag for this slot to trigger concealment */
+ self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 1;
+ errorStatus = SBRDEC_PARSE_ERROR;
+ } else {
+ /* Everything seems to be ok so clear the error flag */
+ self->pSbrElement[elementIndex]->frameErrorFlag[hSbrElement->useFrameSlot] = 0;
+ }
+
+ if (!stereo) {
+ /* Turn coupling off explicitely to avoid access to absent right frame data
+ that might occur with corrupt bitstreams. */
+ hFrameDataLeft->coupling = COUPLING_OFF;
+ }
+
+bail:
+ if (errorStatus == SBRDEC_OK) {
+ if (headerStatus == HEADER_NOT_PRESENT) {
+ /* Use the old header for this frame */
+ hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot;
+ } else {
+ /* Use the new header for this frame */
+ hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = thisHdrSlot;
+ }
+
+ /* Move frame pointer to the next slot which is up to be decoded/applied next */
+ hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
+ }
+
+ *count -= startPos - FDKgetValidBits(hBs);
+
+ return errorStatus;
+}
+
+
+/**
+ * \brief Render one SBR element into time domain signal.
+ * \param self SBR decoder handle
+ * \param timeData pointer to output buffer
+ * \param interleaved flag indicating interleaved channel output
+ * \param channelMapping pointer to UCHAR array where next 2 channel offsets are stored.
+ * \param elementIndex enumerating index of the SBR element to render.
+ * \param numInChannels number of channels from core coder (reading stride).
+ * \param numOutChannels pointer to a location to return number of output channels.
+ * \param psPossible flag indicating if PS is possible or not.
+ * \return SBRDEC_OK if successfull, else error code
+ */
+static SBR_ERROR
+sbrDecoder_DecodeElement (
+ HANDLE_SBRDECODER self,
+ INT_PCM *timeData,
+ const int interleaved,
+ const UCHAR *channelMapping,
+ const int elementIndex,
+ const int numInChannels,
+ int *numOutChannels,
+ const int psPossible
+ )
+{
+ SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex];
+ HANDLE_SBR_CHANNEL *pSbrChannel = self->pSbrElement[elementIndex]->pSbrChannel;
+ HANDLE_SBR_HEADER_DATA hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
+ HANDLE_PS_DEC h_ps_d = self->hParametricStereoDec;
+
+ /* get memory for frame data from scratch */
+ SBR_FRAME_DATA *hFrameDataLeft = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+ SBR_FRAME_DATA *hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+
+ SBR_ERROR errorStatus = SBRDEC_OK;
+
+
+ INT strideIn, strideOut, offset0, offset1;
+ INT codecFrameSize = self->codecFrameSize;
+
+ int stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
+ int numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
+
+ /* Update the header error flag */
+ hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot];
+
+ /*
+ Prepare filterbank for upsampling if no valid bit stream data is available.
+ */
+ if ( hSbrHeader->syncState == SBR_NOT_INITIALIZED )
+ {
+ errorStatus = initHeaderData(
+ hSbrHeader,
+ self->sampleRateIn,
+ self->sampleRateOut,
+ codecFrameSize,
+ self->flags
+ );
+
+ if (errorStatus != SBRDEC_OK) {
+ return errorStatus;
+ }
+
+ hSbrHeader->syncState = UPSAMPLING;
+
+ errorStatus = sbrDecoder_HeaderUpdate(
+ self,
+ hSbrHeader,
+ HEADER_NOT_PRESENT,
+ pSbrChannel,
+ hSbrElement->nChannels
+ );
+
+ if (errorStatus != SBRDEC_OK) {
+ hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+ return errorStatus;
+ }
+ }
+
+ /* reset */
+ if (hSbrHeader->status & SBRDEC_HDR_STAT_RESET) {
+ int ch;
+ for (ch = 0 ; ch < numElementChannels; ch++) {
+ SBR_ERROR errorStatusTmp = SBRDEC_OK;
+
+ errorStatusTmp = resetSbrDec (
+ &pSbrChannel[ch]->SbrDec,
+ hSbrHeader,
+ &pSbrChannel[ch]->prevFrameData,
+ self->flags & SBRDEC_LOW_POWER,
+ self->synDownsampleFac
+ );
+
+ if (errorStatusTmp != SBRDEC_OK) {
+ errorStatus = errorStatusTmp;
+ }
+ }
+ hSbrHeader->status &= ~SBRDEC_HDR_STAT_RESET;
+ }
+
+ /* decoding */
+ if ( (hSbrHeader->syncState == SBR_ACTIVE)
+ || ((hSbrHeader->syncState == SBR_HEADER) && (hSbrHeader->frameErrorFlag == 0)) )
+ {
+ errorStatus = SBRDEC_OK;
+
+ decodeSbrData (hSbrHeader,
+ hFrameDataLeft,
+ &pSbrChannel[0]->prevFrameData,
+ (stereo) ? hFrameDataRight : NULL,
+ (stereo) ? &pSbrChannel[1]->prevFrameData : NULL);
+
+
+ /* Now we have a full parameter set and can do parameter
+ based concealment instead of plain upsampling. */
+ hSbrHeader->syncState = SBR_ACTIVE;
+ }
+
+ /* decode PS data if available */
+ if (h_ps_d != NULL && psPossible) {
+ int applyPs = 1;
+
+ /* define which frame delay line slot to process */
+ h_ps_d->processSlot = hSbrElement->useFrameSlot;
+
+ applyPs = DecodePs(h_ps_d, hSbrHeader->frameErrorFlag);
+ self->flags |= (applyPs) ? SBRDEC_PS_DECODED : 0;
+ }
+
+ /* Set strides for reading and writing */
+ if (interleaved) {
+ strideIn = numInChannels;
+ if ( psPossible )
+ strideOut = (numInChannels < 2) ? 2 : numInChannels;
+ else
+ strideOut = numInChannels;
+ offset0 = channelMapping[0];
+ offset1 = channelMapping[1];
+ } else {
+ strideIn = 1;
+ strideOut = 1;
+ offset0 = channelMapping[0]*2*codecFrameSize;
+ offset1 = channelMapping[1]*2*codecFrameSize;
+ }
+
+ /* use same buffers for left and right channel and apply PS per timeslot */
+ /* Process left channel */
+//FDKprintf("self->codecFrameSize %d\t%d\n",self->codecFrameSize,self->sampleRateIn);
+ sbr_dec (&pSbrChannel[0]->SbrDec,
+ timeData + offset0,
+ timeData + offset0,
+ &pSbrChannel[1]->SbrDec,
+ timeData + offset1,
+ strideIn,
+ strideOut,
+ hSbrHeader,
+ hFrameDataLeft,
+ &pSbrChannel[0]->prevFrameData,
+ (hSbrHeader->syncState == SBR_ACTIVE),
+ h_ps_d,
+ self->flags
+ );
+
+ if (stereo) {
+ /* Process right channel */
+ sbr_dec (&pSbrChannel[1]->SbrDec,
+ timeData + offset1,
+ timeData + offset1,
+ NULL,
+ NULL,
+ strideIn,
+ strideOut,
+ hSbrHeader,
+ hFrameDataRight,
+ &pSbrChannel[1]->prevFrameData,
+ (hSbrHeader->syncState == SBR_ACTIVE),
+ NULL,
+ self->flags
+ );
+ }
+
+ if (h_ps_d != NULL) {
+ /* save PS status for next run */
+ h_ps_d->psDecodedPrv = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0 ;
+ }
+
+ if ( psPossible
+ )
+ {
+ FDK_ASSERT(strideOut > 1);
+ if ( !(self->flags & SBRDEC_PS_DECODED) ) {
+ /* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */
+ /* So copy left channel to right channel. */
+ if (interleaved) {
+ INT_PCM *ptr;
+ INT i;
+ FDK_ASSERT(strideOut == 2);
+
+ ptr = timeData;
+ for (i = codecFrameSize; i--; )
+ {
+ INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */
+ tmp = *ptr++; *ptr++ = tmp;
+ tmp = *ptr++; *ptr++ = tmp;
+ }
+ } else {
+ FDKmemcpy( timeData+2*codecFrameSize, timeData, 2*codecFrameSize*sizeof(INT_PCM) );
+ }
+ }
+ *numOutChannels = 2; /* Output minimum two channels when PS is enabled. */
+ }
+
+ return errorStatus;
+}
+
+
+SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER self,
+ INT_PCM *timeData,
+ int *numChannels,
+ int *sampleRate,
+ const UCHAR channelMapping[(6)],
+ const int interleaved,
+ const int coreDecodedOk,
+ UCHAR *psDecoded )
+{
+ SBR_ERROR errorStatus = SBRDEC_OK;
+
+ int psPossible = 0;
+ int sbrElementNum;
+ int numCoreChannels = *numChannels;
+ int numSbrChannels = 0;
+
+ psPossible = *psDecoded;
+
+ if (self->numSbrElements < 1) {
+ /* exit immediately to avoid access violations */
+ return SBRDEC_CREATE_ERROR;
+ }
+
+ /* Sanity check of allocated SBR elements. */
+ for (sbrElementNum=0; sbrElementNum<self->numSbrElements; sbrElementNum++) {
+ if (self->pSbrElement[sbrElementNum] == NULL) {
+ return SBRDEC_CREATE_ERROR;
+ }
+ }
+
+ if (self->numSbrElements != 1 || self->pSbrElement[0]->elementID != ID_SCE) {
+ psPossible = 0;
+ }
+
+
+ /* In case of non-interleaved time domain data and upsampling, make room for bigger SBR output. */
+ if (self->synDownsampleFac == 1 && interleaved == 0) {
+ int c, outputFrameSize;
+
+ outputFrameSize =
+ self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_channels
+ * self->pSbrElement[0]->pSbrChannel[0]->SbrDec.SynthesisQMF.no_col;
+
+ for (c=numCoreChannels-1; c>0; c--) {
+ FDKmemmove(timeData + c*outputFrameSize, timeData + c*self->codecFrameSize , self->codecFrameSize*sizeof(INT_PCM));
+ }
+ }
+
+
+ /* Make sure that even if no SBR data was found/parsed *psDecoded is returned 1 if psPossible was 0. */
+ if (psPossible == 0) {
+ self->flags &= ~SBRDEC_PS_DECODED;
+ }
+
+ /* Loop over SBR elements */
+ for (sbrElementNum = 0; sbrElementNum<self->numSbrElements; sbrElementNum++)
+ {
+ int numElementChan;
+
+ if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) {
+ errorStatus = SBRDEC_UNSUPPORTED_CONFIG;
+ goto bail;
+ }
+
+ numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1;
+
+ /* If core signal is bad then force upsampling */
+ if ( ! coreDecodedOk ) {
+ self->pSbrElement[sbrElementNum]->frameErrorFlag[self->pSbrElement[sbrElementNum]->useFrameSlot] = 1;
+ }
+
+ errorStatus = sbrDecoder_DecodeElement (
+ self,
+ timeData,
+ interleaved,
+ channelMapping,
+ sbrElementNum,
+ numCoreChannels,
+ &numElementChan,
+ psPossible
+ );
+
+ if (errorStatus != SBRDEC_OK) {
+ goto bail;
+ }
+
+ numSbrChannels += numElementChan;
+ channelMapping += numElementChan;
+
+ if (numSbrChannels >= numCoreChannels) {
+ break;
+ }
+ }
+
+ /* Update numChannels and samplerate */
+ *numChannels = numSbrChannels;
+ *sampleRate = self->sampleRateOut;
+ *psDecoded = (self->flags & SBRDEC_PS_DECODED) ? 1 : 0;
+
+
+
+bail:
+
+ return errorStatus;
+}
+
+
+SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *pSelf )
+{
+ HANDLE_SBRDECODER self = *pSelf;
+ int i;
+
+ if (self != NULL)
+ {
+ if (self->hParametricStereoDec != NULL) {
+ DeletePsDec ( &self->hParametricStereoDec );
+ }
+
+ if (self->workBuffer1 != NULL) {
+ FreeRam_SbrDecWorkBuffer1(&self->workBuffer1);
+ }
+ if (self->workBuffer2 != NULL) {
+ FreeRam_SbrDecWorkBuffer2(&self->workBuffer2);
+ }
+
+ for (i = 0; i < (4); i++) {
+ sbrDecoder_DestroyElement( self, i );
+ }
+
+ FreeRam_SbrDecoder(pSelf);
+ }
+
+ return SBRDEC_OK;
+}
+
+
+INT sbrDecoder_GetLibInfo( LIB_INFO *info )
+{
+ int i;
+
+ if (info == NULL) {
+ return -1;
+ }
+
+ /* search for next free tab */
+ for (i = 0; i < FDK_MODULE_LAST; i++) {
+ if (info[i].module_id == FDK_NONE)
+ break;
+ }
+ if (i == FDK_MODULE_LAST)
+ return -1;
+ info += i;
+
+ info->module_id = FDK_SBRDEC;
+ info->version = LIB_VERSION(SBRDECODER_LIB_VL0, SBRDECODER_LIB_VL1, SBRDECODER_LIB_VL2);
+ LIB_VERSION_STRING(info);
+ info->build_date = (char *)SBRDECODER_LIB_BUILD_DATE;
+ info->build_time = (char *)SBRDECODER_LIB_BUILD_TIME;
+ info->title = (char *)SBRDECODER_LIB_TITLE;
+
+ /* Set flags */
+ info->flags = 0
+ | CAPF_SBR_HQ
+ | CAPF_SBR_LP
+ | CAPF_SBR_PS_MPEG
+ | CAPF_SBR_CONCEALMENT
+ | CAPF_SBR_DRC
+ ;
+ /* End of flags */
+
+ return 0;
+}
+
diff --git a/libSBRdec/src/transcendent.h b/libSBRdec/src/transcendent.h
new file mode 100644
index 0000000..f0ee21e
--- /dev/null
+++ b/libSBRdec/src/transcendent.h
@@ -0,0 +1,355 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+ \brief FDK Fixed Point Arithmetic Library Interface
+*/
+
+#ifndef __TRANSCENDENT_H
+#define __TRANSCENDENT_H
+
+#include "sbrdecoder.h"
+#include "sbr_rom.h"
+
+/************************************************************************/
+/*!
+ \brief Get number of octaves between frequencies a and b
+
+ The Result is scaled with 1/8.
+ The valid range for a and b is 1 to LOG_DUALIS_TABLE_SIZE.
+
+ \return ld(a/b) / 8
+*/
+/************************************************************************/
+static inline FIXP_SGL FDK_getNumOctavesDiv8(INT a, /*!< lower band */
+ INT b) /*!< upper band */
+{
+ return ( (SHORT)((LONG)(CalcLdInt(b) - CalcLdInt(a))>>(FRACT_BITS-3)) );
+}
+
+
+/************************************************************************/
+/*!
+ \brief Add two values given by mantissa and exponent.
+
+ Mantissas are in fract format with values between 0 and 1. <br>
+ The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br>
+*/
+/************************************************************************/
+inline void FDK_add_MantExp(FIXP_SGL a_m, /*!< Mantissa of 1st operand a */
+ SCHAR a_e, /*!< Exponent of 1st operand a */
+ FIXP_SGL b_m, /*!< Mantissa of 2nd operand b */
+ SCHAR b_e, /*!< Exponent of 2nd operand b */
+ FIXP_SGL *ptrSum_m, /*!< Mantissa of result */
+ SCHAR *ptrSum_e) /*!< Exponent of result */
+{
+ FIXP_DBL accu;
+ int shift;
+ int shiftAbs;
+
+ FIXP_DBL shiftedMantissa;
+ FIXP_DBL otherMantissa;
+
+ /* Equalize exponents of the summands.
+ For the smaller summand, the exponent is adapted and
+ for compensation, the mantissa is shifted right. */
+
+ shift = (int)(a_e - b_e);
+
+ shiftAbs = (shift>0)? shift : -shift;
+ shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1;
+ shiftedMantissa = (shift>0)? (FX_SGL2FX_DBL(b_m) >> shiftAbs) : (FX_SGL2FX_DBL(a_m) >> shiftAbs);
+ otherMantissa = (shift>0)? FX_SGL2FX_DBL(a_m) : FX_SGL2FX_DBL(b_m);
+ *ptrSum_e = (shift>0)? a_e : b_e;
+
+ accu = (shiftedMantissa >> 1) + (otherMantissa >> 1);
+ /* shift by 1 bit to avoid overflow */
+
+ if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) )
+ *ptrSum_e += 1;
+ else
+ accu = (shiftedMantissa + otherMantissa);
+
+ *ptrSum_m = FX_DBL2FX_SGL(accu);
+
+}
+
+inline void FDK_add_MantExp(FIXP_DBL a, /*!< Mantissa of 1st operand a */
+ SCHAR a_e, /*!< Exponent of 1st operand a */
+ FIXP_DBL b, /*!< Mantissa of 2nd operand b */
+ SCHAR b_e, /*!< Exponent of 2nd operand b */
+ FIXP_DBL *ptrSum, /*!< Mantissa of result */
+ SCHAR *ptrSum_e) /*!< Exponent of result */
+{
+ FIXP_DBL accu;
+ int shift;
+ int shiftAbs;
+
+ FIXP_DBL shiftedMantissa;
+ FIXP_DBL otherMantissa;
+
+ /* Equalize exponents of the summands.
+ For the smaller summand, the exponent is adapted and
+ for compensation, the mantissa is shifted right. */
+
+ shift = (int)(a_e - b_e);
+
+ shiftAbs = (shift>0)? shift : -shift;
+ shiftAbs = (shiftAbs < DFRACT_BITS-1)? shiftAbs : DFRACT_BITS-1;
+ shiftedMantissa = (shift>0)? (b >> shiftAbs) : (a >> shiftAbs);
+ otherMantissa = (shift>0)? a : b;
+ *ptrSum_e = (shift>0)? a_e : b_e;
+
+ accu = (shiftedMantissa >> 1) + (otherMantissa >> 1);
+ /* shift by 1 bit to avoid overflow */
+
+ if ( (accu >= (FL2FXCONST_DBL(0.5f) - (FIXP_DBL)1)) || (accu <= FL2FXCONST_DBL(-0.5f)) )
+ *ptrSum_e += 1;
+ else
+ accu = (shiftedMantissa + otherMantissa);
+
+ *ptrSum = accu;
+
+}
+
+/************************************************************************/
+/*!
+ \brief Divide two values given by mantissa and exponent.
+
+ Mantissas are in fract format with values between 0 and 1. <br>
+ The base for exponents is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br>
+
+ For performance reasons, the division is based on a table lookup
+ which limits accuracy.
+*/
+/************************************************************************/
+static inline void FDK_divide_MantExp(FIXP_SGL a_m, /*!< Mantissa of dividend a */
+ SCHAR a_e, /*!< Exponent of dividend a */
+ FIXP_SGL b_m, /*!< Mantissa of divisor b */
+ SCHAR b_e, /*!< Exponent of divisor b */
+ FIXP_SGL *ptrResult_m, /*!< Mantissa of quotient a/b */
+ SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */
+
+{
+ int preShift, postShift, index, shift;
+ FIXP_DBL ratio_m;
+ FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f);
+
+ preShift = CntLeadingZeros(FX_SGL2FX_DBL(b_m));
+
+ /*
+ Shift b into the range from 0..INV_TABLE_SIZE-1,
+
+ E.g. 10 bits must be skipped for INV_TABLE_BITS 8:
+ - leave 8 bits as index for table
+ - skip sign bit,
+ - skip first bit of mantissa, because this is always the same (>0.5)
+
+ We are dealing with energies, so we need not care
+ about negative numbers
+ */
+
+ /*
+ The first interval has half width so the lowest bit of the index is
+ needed for a doubled resolution.
+ */
+ shift = (FRACT_BITS - 2 - INV_TABLE_BITS - preShift);
+
+ index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift;
+
+
+ /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */
+ index &= (1 << (INV_TABLE_BITS+1)) - 1;
+
+ /* Remove offset of half an interval */
+ index--;
+
+ /* Now the lowest bit is shifted out */
+ index = index >> 1;
+
+ /* Fetch inversed mantissa from table: */
+ bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index];
+
+ /* Multiply a with the inverse of b: */
+ ratio_m = (index<0)? FX_SGL2FX_DBL(a_m >> 1) : fMultDiv2(bInv_m,a_m);
+
+ postShift = CntLeadingZeros(ratio_m)-1;
+
+ *ptrResult_m = FX_DBL2FX_SGL(ratio_m << postShift);
+ *ptrResult_e = a_e - b_e + 1 + preShift - postShift;
+}
+
+static inline void FDK_divide_MantExp(FIXP_DBL a_m, /*!< Mantissa of dividend a */
+ SCHAR a_e, /*!< Exponent of dividend a */
+ FIXP_DBL b_m, /*!< Mantissa of divisor b */
+ SCHAR b_e, /*!< Exponent of divisor b */
+ FIXP_DBL *ptrResult_m, /*!< Mantissa of quotient a/b */
+ SCHAR *ptrResult_e) /*!< Exponent of quotient a/b */
+
+{
+ int preShift, postShift, index, shift;
+ FIXP_DBL ratio_m;
+ FIXP_SGL bInv_m = FL2FXCONST_SGL(0.0f);
+
+ preShift = CntLeadingZeros(b_m);
+
+ /*
+ Shift b into the range from 0..INV_TABLE_SIZE-1,
+
+ E.g. 10 bits must be skipped for INV_TABLE_BITS 8:
+ - leave 8 bits as index for table
+ - skip sign bit,
+ - skip first bit of mantissa, because this is always the same (>0.5)
+
+ We are dealing with energies, so we need not care
+ about negative numbers
+ */
+
+ /*
+ The first interval has half width so the lowest bit of the index is
+ needed for a doubled resolution.
+ */
+ shift = (DFRACT_BITS - 2 - INV_TABLE_BITS - preShift);
+
+ index = (shift<0)? (LONG)b_m << (-shift) : (LONG)b_m >> shift;
+
+
+ /* The index has INV_TABLE_BITS +1 valid bits here. Clear the other bits. */
+ index &= (1 << (INV_TABLE_BITS+1)) - 1;
+
+ /* Remove offset of half an interval */
+ index--;
+
+ /* Now the lowest bit is shifted out */
+ index = index >> 1;
+
+ /* Fetch inversed mantissa from table: */
+ bInv_m = (index<0)? bInv_m : FDK_sbrDecoder_invTable[index];
+
+ /* Multiply a with the inverse of b: */
+ ratio_m = (index<0)? (a_m >> 1) : fMultDiv2(bInv_m,a_m);
+
+ postShift = CntLeadingZeros(ratio_m)-1;
+
+ *ptrResult_m = ratio_m << postShift;
+ *ptrResult_e = a_e - b_e + 1 + preShift - postShift;
+}
+
+/*!
+ \brief Calculate the squareroot of a number given by mantissa and exponent
+
+ Mantissa is in fract format with values between 0 and 1. <br>
+ The base for the exponent is 2. Example: \f$ a = a\_m * 2^{a\_e} \f$<br>
+ The operand is addressed via pointers and will be overwritten with the result.
+
+ For performance reasons, the square root is based on a table lookup
+ which limits accuracy.
+*/
+static inline void FDK_sqrt_MantExp(FIXP_DBL *mantissa, /*!< Pointer to mantissa */
+ SCHAR *exponent,
+ const SCHAR *destScale)
+{
+ FIXP_DBL input_m = *mantissa;
+ int input_e = (int) *exponent;
+ FIXP_DBL result = FL2FXCONST_DBL(0.0f);
+ int result_e = -FRACT_BITS;
+
+ /* Call lookup square root, which does internally normalization. */
+ result = sqrtFixp_lookup(input_m, &input_e);
+ result_e = input_e;
+
+ /* Write result */
+ if (exponent==destScale) {
+ *mantissa = result;
+ *exponent = result_e;
+ } else {
+ int shift = result_e - *destScale;
+ *mantissa = (shift>=0) ? result << (INT)fixMin(DFRACT_BITS-1,shift)
+ : result >> (INT)fixMin(DFRACT_BITS-1,-shift);
+ *exponent = *destScale;
+ }
+}
+
+
+#endif