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+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/**************************** SBR encoder library ******************************
+
+ 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 enum codecType {
+ CODEC_AAC = 0,
+ CODEC_AACLD = 1,
+ CODEC_UNSPECIFIED = 99
+} CODEC_TYPE;
+
+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,
+ SBR_SYNTAX_ELD_REDUCED_DELAY = 0x0010
+};
+
+typedef enum { FREQ_RES_LOW = 0, FREQ_RES_HIGH } FREQ_RES;
+
+typedef struct {
+ CODEC_TYPE coreCoder; /*!< LC or ELD */
+ UINT bitrateFrom; /*!< inclusive */
+ UINT bitrateTo; /*!< exclusive */
+
+ UINT 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 downSampleFactor; /*!< Sampling rate relation between the SBR and the core
+ encoder. */
+ FREQ_RES freq_res_fixfix[2]; /*!< Frequency resolution of envelopes in frame
+ class FIXFIX, for non-split case and split
+ case */
+ UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient
+ frames: low (0) or variable (1) */
+
+ /*
+ 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. */
+
+ /*
+ 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;
+ FIXP_DBL threshold_AmpRes_FF_m;
+ SCHAR threshold_AmpRes_FF_e;
+} sbrConfiguration, *sbrConfigurationPtr;
+
+typedef struct SBR_CONFIG_DATA {
+ 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 coeffs 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;
+ FIXP_DBL thresholdAmpResFF_m;
+ SCHAR thresholdAmpResFF_e;
+} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
+
+typedef struct {
+ MP4_ELEMENT_ID elType;
+ INT bitRate;
+ int instanceTag;
+ UCHAR fParametricStereo;
+ UCHAR fDualMono; /**< This flags allows to disable coupling in sbr channel
+ pair element */
+ 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 bitrate to specified desired
+ * bitrate 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 aot 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 Check whether downsampled SBR single rate is possible
+ * with given audio object type.
+ * \param aot The Audio object type.
+ * \return 0 when downsampled SBR is not possible,
+ * 1 when downsampled SBR is possible.
+ */
+UINT sbrEncoder_IsSingleRatePossible(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 inputBufferBufSize Buffer offset of one channel (frameSize + delay)
+ * \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 downSampleFactor Input: Relation between SBR and core coder sampling
+ * rate;
+ * \param frameLength Input: Encoder frameLength. output core encoder
+ * frameLength.
+ * \param aot Input: AOT..
+ * \param delay Input: core encoder delay. Output: total delay
+ * because of SBR.
+ * \param transformFactor The core encoder transform factor (blockswitching).
+ * \param headerPeriod Repetition rate of the SBR header:
+ * - (-1) means intern configuration.
+ * - (1-10) corresponds to header repetition rate in
+ * frames.
+ * \return 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder,
+ SBR_ELEMENT_INFO elInfo[(8)], int noElements,
+ INT_PCM *inputBuffer, UINT inputBufferBufSize,
+ INT *coreBandwidth, INT *inputBufferOffset,
+ INT *numChannels, const UINT syntaxFlags, INT *sampleRate,
+ UINT *downSampleFactor, INT *frameLength,
+ AUDIO_OBJECT_TYPE aot, int *delay, int transformFactor,
+ const int headerPeriod, 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.
+ * \param timeBufferBufSIze buffer size for one channel
+ * \return 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc, INT_PCM *timeBuffer,
+ UINT timeBufferBufSIze);
+
+/**
+ * \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, not interleaved.
+ * \param timeInStride Channel offset 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 samplesBufSize, UINT sbrDataBits[(8)],
+ UCHAR sbrData[(8)][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(HANDLE_SBR_ENCODER sbrEncoder,
+ HANDLE_FDK_BITSTREAM hBs, INT element_index,
+ int fSendHeaders);
+
+/**
+ * \brief Request to write SBR header.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief Request if last sbr payload contains an SBR header.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return 1 contains sbr header, 0 without sbr header.
+ */
+INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief SBR header delay in frames.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return Delay in frames, -1 on failure.
+ */
+INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief Bitstrem delay in SBR frames.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return Delay in frames, -1 on failure.
+ */
+INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \brief Prepare SBR payload for SAP.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return 0 on success, and non-zero if failed.
+ */
+INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder);
+
+/**
+ * \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 Delay caused by the SBR decoder.
+ * \param hSbrEncoder SBR encoder handle.
+ * \return Delay.
+ */
+INT sbrEncoder_GetSbrDecDelay(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 */