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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/***************************  Fraunhofer IIS ***********************
+
+   Author(s):
+   Description: SBR encoder top level processing prototype
+
+******************************************************************************/
+
+#ifndef __SBR_ENCODER_H
+#define __SBR_ENCODER_H
+
+#include "common_fix.h"
+#include "FDK_audio.h"
+
+#include "FDK_bitstream.h"
+
+/* core coder helpers */
+#define MAX_TRANS_FAC         8
+#define MAX_CODEC_FRAME_RATIO 2
+#define MAX_PAYLOAD_SIZE    256
+
+typedef struct
+{
+  INT bitRate;
+  INT nChannels;
+  INT sampleFreq;
+  INT transFac;
+  INT standardBitrate;
+} CODEC_PARAM;
+
+typedef enum
+{
+  SBR_MONO,
+  SBR_LEFT_RIGHT,
+  SBR_COUPLING,
+  SBR_SWITCH_LRC
+} SBR_STEREO_MODE;
+
+/* bitstream syntax flags */
+enum
+{
+  SBR_SYNTAX_LOW_DELAY = 0x0001,
+  SBR_SYNTAX_SCALABLE  = 0x0002,
+  SBR_SYNTAX_CRC       = 0x0004,
+  SBR_SYNTAX_DRM_CRC   = 0x0008
+};
+
+typedef struct
+{
+  UINT             bitrateFrom;      /*!< inclusive */
+  UINT             bitrateTo;        /*!< exclusive */
+
+  USHORT           sampleRate;       /*!<   */
+  UCHAR            numChannels;      /*!<   */
+
+  UCHAR            startFreq;        /*!< bs_start_freq */
+  UCHAR            startFreqSpeech;  /*!< bs_start_freq for speech config flag */
+  UCHAR            stopFreq;         /*!< bs_stop_freq */
+  UCHAR            stopFreqSpeech;   /*!< bs_stop_freq for speech config flag */
+
+  UCHAR            numNoiseBands;    /*!<   */
+  UCHAR            noiseFloorOffset; /*!<   */
+  SCHAR            noiseMaxLevel;    /*!<   */
+  SBR_STEREO_MODE  stereoMode;       /*!<   */
+  UCHAR            freqScale;        /*!<   */
+} sbrTuningTable_t;
+
+typedef struct sbrConfiguration
+{
+  /*
+     core coder dependent configurations
+  */
+  CODEC_PARAM codecSettings;  /*!< Core coder settings. To be set from core coder. */
+  INT SendHeaderDataTime;     /*!< SBR header send update frequency in ms. */
+  INT useWaveCoding;          /*!< Flag: usage of wavecoding tool. */
+  INT crcSbr;                 /*!< Flag: usage of SBR-CRC. */
+  INT dynBwSupported;         /*!< Flag: support for dynamic bandwidth in this combination. */
+  INT parametricCoding;       /*!< Flag: usage of parametric coding tool. */
+  int freq_res_fixfix[3];     /*!< Frequency resolution of envelopes in frame class FIXFIX
+                                 0=1 Env; 1=2 Env; 2=4 Env; */
+  /*
+     core coder dependent tuning parameters
+  */
+  INT tran_thr;             /*!< SBR transient detector threshold (* 100). */
+  INT noiseFloorOffset;     /*!< Noise floor offset.      */
+  UINT useSpeechConfig;     /*!< Flag: adapt tuning parameters according to speech. */
+
+
+
+  /*
+     core coder independent configurations
+  */
+  INT sbrFrameSize;           /*!< SBR frame size in samples. Will be calculated from core coder settings. */
+  INT sbr_data_extra;         /*!< Flag usage of data extra. */
+  INT amp_res;                /*!< Amplitude resolution. */
+  INT ana_max_level;          /*!< Noise insertion maximum level. */
+  INT tran_fc;                /*!< Transient detector start frequency. */
+  INT tran_det_mode;          /*!< Transient detector mode. */
+  INT spread;                 /*!< Flag: usage of SBR spread. */
+  INT stat;                   /*!< Flag: usage of static framing. */
+  INT e;                      /*!< Number of envelopes when static framing is chosen. */
+  SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */
+  INT deltaTAcrossFrames;     /*!< Flag: allow time-delta coding. */
+  FIXP_DBL dF_edge_1stEnv;    /*!< Extra fraction delta-F coding is allowed to be more expensive. */
+  FIXP_DBL dF_edge_incr;      /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */
+  INT sbr_invf_mode;          /*!< Inverse filtering mode. */
+  INT sbr_xpos_mode;          /*!< Transposer mode. */
+  INT sbr_xpos_ctrl;          /*!< Transposer control. */
+  INT sbr_xpos_level;         /*!< Transposer 3rd order level. */
+  INT startFreq;              /*!< The start frequency table index. */
+  INT stopFreq;               /*!< The stop frequency table index. */
+  INT useSaPan;               /*!< Flag: usage of SAPAN stereo. */
+  INT dynBwEnabled;           /*!< Flag: usage of dynamic bandwidth. */
+  INT bParametricStereo;      /*!< Flag: usage of parametric stereo coding tool. */
+  INT bDownSampledSbr;        /*!< Signal downsampled SBR is used. */
+
+  /*
+     header_extra1 configuration
+  */
+  UCHAR freqScale;            /*!< Frequency grouping. */
+  INT alterScale;             /*!< Scale resolution. */
+  INT sbr_noise_bands;        /*!< Number of noise bands. */
+
+
+  /*
+     header_extra2 configuration
+  */
+  INT sbr_limiter_bands;      /*!< Number of limiter bands. */
+  INT sbr_limiter_gains;      /*!< Gain of limiter. */
+  INT sbr_interpol_freq;      /*!< Flag: use interpolation in freq. direction. */
+  INT sbr_smoothing_length;   /*!< Flag: choose length 4 or 0 (=on, off). */
+  UCHAR init_amp_res_FF;
+} sbrConfiguration, *sbrConfigurationPtr ;
+
+typedef struct
+{
+  UINT sbrSyntaxFlags;                  /**< SBR syntax flags derived from AOT. */
+  INT nChannels;                        /**< Number of channels.  */
+
+  INT nSfb[2];                          /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */
+  INT num_Master;                       /**< Number of elements in v_k_master. */
+  INT sampleFreq;                       /**< SBR sampling frequency. */
+  INT frameSize;
+  INT xOverFreq;                        /**< The SBR start frequency. */
+  INT dynXOverFreq;                     /**< Used crossover frequency when dynamic bandwidth is enabled. */
+  INT noQmfBands;                       /**< Number of QMF frequency bands. */
+  INT noQmfSlots;                       /**< Number of QMF slots. */
+
+  UCHAR *freqBandTable[2];              /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeefs actually needed for lowres. */
+  UCHAR *v_k_master;                    /**< Master BandTable where freqBandTable is derived from. */
+
+
+  SBR_STEREO_MODE stereoMode;
+  INT noEnvChannels;                    /**< Number of envelope channels. */
+
+  INT useWaveCoding;                    /**< Flag indicates whether to use wave coding at all.      */
+  INT useParametricCoding;              /**< Flag indicates whether to use para coding at all.      */
+  INT xposCtrlSwitch;                   /**< Flag indicates whether to switch xpos ctrl on the fly. */
+  INT switchTransposers;                /**< Flag indicates whether to switch xpos on the fly .     */
+  UCHAR initAmpResFF;
+} SBR_CONFIG_DATA;
+
+typedef SBR_CONFIG_DATA *HANDLE_SBR_CONFIG_DATA;
+
+typedef struct {
+  MP4_ELEMENT_ID elType;
+  INT bitRate;
+  int instanceTag;
+  UCHAR fParametricStereo;
+  UCHAR nChannelsInEl;
+  UCHAR ChannelIndex[2];
+} SBR_ELEMENT_INFO;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER;
+
+/**
+ * \brief  Get the max required input buffer size including delay balancing space
+ *         for N audio channels.
+ * \param noChannels  Number of audio channels.
+ * \return            Max required input buffer size in bytes.
+ */
+INT sbrEncoder_GetInBufferSize(int noChannels);
+
+INT sbrEncoder_Open(
+        HANDLE_SBR_ENCODER  *phSbrEncoder,
+        INT                  nElements,
+        INT                  nChannels,
+        INT                  supportPS
+        );
+
+/**
+ * \brief get closest working bit rate to specified desired bit rate for a single SBR element
+ * \param bitRate the desired target bit rate
+ * \param numChannels the amount of audio channels
+ * \param coreSampleRate the sample rate of the core coder
+ * \param the current Audio Object Type
+ * \return closest working bit rate to bitRate value
+ */
+UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot);
+
+/**
+ * \brief                  Initialize SBR Encoder instance.
+ * \param phSbrEncoder     Pointer to a SBR Encoder instance.
+ * \param elInfo           Structure that describes the element/channel arrangement.
+ * \param noElements       Amount of elements described in elInfo.
+ * \param inputBuffer      Pointer to the encoder audio buffer
+ * \param bandwidth        Returns the core audio encoder bandwidth (output)
+ * \param bufferOffset     Returns the offset for the audio input data in order to do delay balancing.
+ * \param numChannels      Input: Encoder input channels. output: core encoder channels.
+ * \param sampleRate       Input: Encoder samplerate. output core encoder samplerate.
+ * \param frameLength      Input: Encoder frameLength. output core encoder frameLength.
+ * \param aot              Input: Desired AOT. output AOT to be used after parameter checking.
+ * \param delay            Input: core encoder delay. Output: total delay because of SBR.
+ * \param transformFactor  The core encoder transform factor (blockswitching).
+ * \return                 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder,
+                     SBR_ELEMENT_INFO elInfo[(6)],
+                     int              noElements,
+                     INT_PCM *inputBuffer,
+                     INT   *bandwidth,
+                     INT   *bufferOffset,
+                     INT   *numChannels,
+                     INT   *sampleRate,
+                     INT   *frameLength,
+                     AUDIO_OBJECT_TYPE *aot,
+                     int *delay,
+                     int  transformFactor,
+                     ULONG statesInitFlag
+                    );
+
+/**
+ * \brief             Do delay line buffers housekeeping. To be called after each encoded audio frame.
+ * \param hEnvEnc     SBR Encoder handle.
+ * \param timeBuffer  Pointer to the encoder audio buffer.
+ * \return            0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc,
+                             INT_PCM *timeBuffer
+                            );
+
+/**
+ * \brief               Close SBR encoder instance.
+ * \param phEbrEncoder  Handle of SBR encoder instance to be closed.
+ * \return              void
+ */
+void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder);
+
+/**
+ * \brief               Encode SBR data of one complete audio frame.
+ * \param hEnvEncoder   Handle of SBR encoder instance.
+ * \param samples       Time samples, always interleaved.
+ * \param timeInStride  Channel stride factor of samples buffer.
+ * \param sbrDataBits   Size of SBR payload in bits.
+ * \param sbrData       SBR payload.
+ * \return              0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER  hEnvEncoder,
+                           INT_PCM            *samples,
+                           UINT                timeInStride,
+                           UINT                sbrDataBits[(6)],
+                           UCHAR               sbrData[(6)][MAX_PAYLOAD_SIZE]
+                          );
+
+/**
+ * \brief               Write SBR headers of one SBR element.
+ * \param sbrEncoder    Handle of the SBR encoder instance.
+ * \param hBs           Handle of bit stream handle to write SBR header to.
+ * \param element_index Index of the SBR element which header should be written.
+ * \param fSendHeaders  Flag indicating that the SBR encoder should send more headers in the SBR payload or not.
+ * \return              void
+ */
+void sbrEncoder_GetHeader(SBR_ENCODER   *sbrEncoder,
+                          HANDLE_FDK_BITSTREAM hBs,
+                          INT            element_index,
+                          int            fSendHeaders);
+
+/**
+ * \brief              SBR encoder bitrate estimation.
+ * \param hSbrEncoder  SBR encoder handle.
+ * \return             Estimated bitrate.
+ */
+INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder);
+
+
+/**
+ * \brief              Delay between input data and downsampled output data.
+ * \param hSbrEncoder  SBR encoder handle.
+ * \return             Delay.
+ */
+INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief       Get decoder library version info.
+ * \param info  Pointer to an allocated LIB_INFO struct, where library info is written to.
+ * \return      0 on sucess.
+ */
+INT sbrEncoder_GetLibInfo(LIB_INFO *info);
+
+void sbrPrintRAM(void);
+
+void sbrPrintROM(void);
+
+#ifdef __cplusplus
+        }
+#endif
+
+#endif /* ifndef __SBR_MAIN_H */