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diff --git a/libSYS/include/FDK_audio.h b/libSYS/include/FDK_audio.h new file mode 100644 index 0000000..8e7041d --- /dev/null +++ b/libSYS/include/FDK_audio.h @@ -0,0 +1,612 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 FDK SysLib ********************** + + Author(s): Manuel Jander + +******************************************************************************/ + +/** \file FDK_audio.h + * \brief Global audio struct and constant definitions. + */ + +#ifndef FDK_AUDIO_H +#define FDK_AUDIO_H + +#include "machine_type.h" +#include "genericStds.h" + +#ifdef __cplusplus +extern "C" +{ +#endif + +/** + * File format identifiers. + */ +typedef enum +{ + FF_UNKNOWN = -1, /**< Unknown format. */ + FF_RAW = 0, /**< No container, bit stream data conveyed "as is". */ + + FF_MP4_3GPP = 3, /**< 3GPP file format. */ + FF_MP4_MP4F = 4, /**< MPEG-4 File format. */ + + FF_RAWPACKETS = 5, /**< Proprietary raw packet file. */ + + FF_DRMCT = 12 /**< Digital Radio Mondial (DRM30/DRM+) CT proprietary file format. */ + +} FILE_FORMAT; + +/** + * Transport type identifiers. + */ +typedef enum +{ + TT_UNKNOWN = -1, /**< Unknown format. */ + TT_MP4_RAW = 0, /**< "as is" access units (packet based since there is obviously no sync layer) */ + TT_MP4_ADIF = 1, /**< ADIF bitstream format. */ + TT_MP4_ADTS = 2, /**< ADTS bitstream format. */ + + TT_MP4_LATM_MCP1 = 6, /**< Audio Mux Elements with muxConfigPresent = 1 */ + TT_MP4_LATM_MCP0 = 7, /**< Audio Mux Elements with muxConfigPresent = 0, out of band StreamMuxConfig */ + + TT_MP4_LOAS = 10, /**< Audio Sync Stream. */ + + TT_DRM = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */ + + TT_MP1_L1 = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */ + TT_MP1_L2 = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */ + TT_MP1_L3 = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */ + + TT_RSVD50 = 50 /**< */ + +} TRANSPORT_TYPE; + +/** + * Audio Object Type definitions. + */ +typedef enum +{ + AOT_NONE = -1, + AOT_NULL_OBJECT = 0, + AOT_AAC_MAIN = 1, /**< Main profile */ + AOT_AAC_LC = 2, /**< Low Complexity object */ + AOT_AAC_SSR = 3, + AOT_AAC_LTP = 4, + AOT_SBR = 5, + AOT_AAC_SCAL = 6, + AOT_TWIN_VQ = 7, + AOT_CELP = 8, + AOT_HVXC = 9, + AOT_RSVD_10 = 10, /**< (reserved) */ + AOT_RSVD_11 = 11, /**< (reserved) */ + AOT_TTSI = 12, /**< TTSI Object */ + AOT_MAIN_SYNTH = 13, /**< Main Synthetic object */ + AOT_WAV_TAB_SYNTH = 14, /**< Wavetable Synthesis object */ + AOT_GEN_MIDI = 15, /**< General MIDI object */ + AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */ + AOT_ER_AAC_LC = 17, /**< Error Resilient(ER) AAC Low Complexity */ + AOT_RSVD_18 = 18, /**< (reserved) */ + AOT_ER_AAC_LTP = 19, /**< Error Resilient(ER) AAC LTP object */ + AOT_ER_AAC_SCAL = 20, /**< Error Resilient(ER) AAC Scalable object */ + AOT_ER_TWIN_VQ = 21, /**< Error Resilient(ER) TwinVQ object */ + AOT_ER_BSAC = 22, /**< Error Resilient(ER) BSAC object */ + AOT_ER_AAC_LD = 23, /**< Error Resilient(ER) AAC LowDelay object */ + AOT_ER_CELP = 24, /**< Error Resilient(ER) CELP object */ + AOT_ER_HVXC = 25, /**< Error Resilient(ER) HVXC object */ + AOT_ER_HILN = 26, /**< Error Resilient(ER) HILN object */ + AOT_ER_PARA = 27, /**< Error Resilient(ER) Parametric object */ + AOT_RSVD_28 = 28, /**< might become SSC */ + AOT_PS = 29, /**< PS, Parametric Stereo (includes SBR) */ + AOT_MPEGS = 30, /**< MPEG Surround */ + + AOT_ESCAPE = 31, /**< Signal AOT uses more than 5 bits */ + + AOT_MP3ONMP4_L1 = 32, /**< MPEG-Layer1 in mp4 */ + AOT_MP3ONMP4_L2 = 33, /**< MPEG-Layer2 in mp4 */ + AOT_MP3ONMP4_L3 = 34, /**< MPEG-Layer3 in mp4 */ + AOT_RSVD_35 = 35, /**< might become DST */ + AOT_RSVD_36 = 36, /**< might become ALS */ + AOT_AAC_SLS = 37, /**< AAC + SLS */ + AOT_SLS = 38, /**< SLS */ + AOT_ER_AAC_ELD = 39, /**< AAC Enhanced Low Delay */ + + AOT_USAC = 42, /**< USAC */ + AOT_SAOC = 43, /**< SAOC */ + AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */ + + AOT_RSVD50 = 50, /**< Interim AOT for Rsvd50 */ + + /* Pseudo AOTs */ + AOT_MP2_AAC_MAIN = 128, /**< Virtual AOT MP2 Main profile */ + AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */ + AOT_MP2_AAC_SSR = 130, /**< Virtual AOT MP2 Scalable Sampling Rate profile */ + + AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */ + + AOT_DAB = 134, /**< Virtual AOT for DAB (Layer2 with scalefactor CRC) */ + AOT_DABPLUS_AAC_LC = 135, /**< Virtual AOT for DAB plus AAC-LC */ + AOT_DABPLUS_SBR = 136, /**< Virtual AOT for DAB plus HE-AAC */ + AOT_DABPLUS_PS = 137, /**< Virtual AOT for DAB plus HE-AAC v2 */ + + AOT_PLAIN_MP1 = 140, /**< Virtual AOT for plain mp1 */ + AOT_PLAIN_MP2 = 141, /**< Virtual AOT for plain mp2 */ + AOT_PLAIN_MP3 = 142, /**< Virtual AOT for plain mp3 */ + + AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */ + AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */ + AOT_DRM_MPEG_PS = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */ + AOT_DRM_SURROUND = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */ + + AOT_MP2_PS = 156, /**< Virtual AOT MP2 Low Complexity Profile with SBR and PS */ + + AOT_MPEGS_RESIDUALS = 256 /**< Virtual AOT for MPEG Surround residuals */ + +} AUDIO_OBJECT_TYPE; + +/** Channel Mode ( 1-7 equals MPEG channel configurations, others are arbitrary). */ +typedef enum { + MODE_INVALID = -1, + MODE_UNKNOWN = 0, + MODE_1 = 1, /**< SCE */ + MODE_2 = 2, /**< CPE */ + MODE_1_2 = 3, /**< SCE,CPE */ + MODE_1_2_1 = 4, /**< SCE,CPE,SCE */ + MODE_1_2_2 = 5, /**< SCE,CPE,CPE */ + MODE_1_2_2_1 = 6, /**< SCE,CPE,CPE,LFE */ + MODE_1_2_2_2_1 = 7, /**< SCE,CPE,CPE,CPE,LFE */ + + MODE_1_1 = 16, /**< 2 SCEs (dual mono) */ + MODE_1_1_1_1 = 17, /**< 4 SCEs */ + MODE_1_1_1_1_1_1 = 18, /**< 6 SCEs */ + MODE_1_1_1_1_1_1_1_1 = 19, /**< 8 SCEs */ + MODE_1_1_1_1_1_1_1_1_1_1_1_1 = 20, /**< 12 SCEs */ + + MODE_2_2 = 21, /**< 2 CPEs */ + MODE_2_2_2 = 22, /**< 3 CPEs */ + MODE_2_2_2_2 = 23, /**< 4 CPEs */ + MODE_2_2_2_2_2_2 = 24, /**< 6 CPEs */ + + MODE_2_1 = 30 /**< CPE,SCE (ARIB standard) */ + +} CHANNEL_MODE; + +/** Speaker description tags */ +typedef enum { + ACT_NONE, + ACT_FRONT, + ACT_SIDE, + ACT_BACK, + ACT_LFE, + ACT_FRONT_TOP, + ACT_SIDE_TOP, + ACT_BACK_TOP, + ACT_TOP /* Ts */ +} AUDIO_CHANNEL_TYPE; + +/** + * Audio Codec flags. + */ +#define AC_ER_VCB11 0x000001 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use virtual codebooks */ +#define AC_ER_RVLC 0x000002 /*!< aacSpectralDataResilienceFlag flag (from ASC): 1 means use huffman codeword reordering */ +#define AC_ER_HCR 0x000004 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use virtual codebooks */ +#define AC_SCALABLE 0x000008 /*!< AAC Scalable*/ +#define AC_ELD 0x000010 /*!< AAC-ELD */ +#define AC_LD 0x000020 /*!< AAC-LD */ +#define AC_ER 0x000040 /*!< ER syntax */ +#define AC_BSAC 0x000080 /*!< BSAC */ +#define AC_USAC 0x000100 /*!< USAC */ +#define AC_USAC_TW 0x000200 /*!< USAC time warped filter bank is active */ +#define AC_USAC_NOISE 0x000400 /*!< USAC noise filling is active */ +#define AC_USAC_HBE 0x000800 /*!< USAC harmonic bandwidth extension is active */ +#define AC_RSVD50 0x001000 /*!< Rsvd50 */ +#define AC_SBR_PRESENT 0x002000 /*!< SBR present flag (from ASC) */ +#define AC_SBRCRC 0x004000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */ +#define AC_PS_PRESENT 0x008000 /*!< PS present flag (from ASC or implicit) */ +#define AC_MPS_PRESENT 0x010000 /*!< MPS present flag (from ASC or implicit) */ +#define AC_DRM 0x020000 /*!< DRM bit stream syntax */ +#define AC_INDEP 0x040000 /*!< Independency flag */ +#define AC_MPS_RES 0x080000 /*!< MPS residual individual channel data. */ +#define AC_DAB 0x800000 /*!< DAB bit stream syntax */ +#define AC_LD_MPS 0x01000000 /*!< Low Delay MPS. */ + + +/* CODER_CONFIG::flags */ +#define CC_MPEG_ID 0x00100000 +#define CC_IS_BASELAYER 0x00200000 +#define CC_PROTECTION 0x00400000 +#define CC_SBR 0x00800000 +#define CC_SBRCRC 0x00010000 +#define CC_RVLC 0x01000000 +#define CC_VCB11 0x02000000 +#define CC_HCR 0x04000000 +#define CC_PSEUDO_SURROUND 0x08000000 +#define CC_USAC_NOISE 0x10000000 +#define CC_USAC_TW 0x20000000 +#define CC_USAC_HBE 0x40000000 + +/** Generic audio coder configuration structure. */ +typedef struct { + AUDIO_OBJECT_TYPE aot; /**< Audio Object Type (AOT). */ + AUDIO_OBJECT_TYPE extAOT; /**< Extension Audio Object Type (SBR). */ + CHANNEL_MODE channelMode; /**< Channel mode. */ + INT samplingRate; /**< Sampling rate. */ + INT extSamplingRate; /**< Extended samplerate (SBR). */ + INT bitRate; /**< Average bitrate. */ + int samplesPerFrame; /**< Number of PCM samples per codec frame and audio channel. */ + int noChannels; /**< Number of audio channels. */ + int bitsFrame; + int nSubFrames; /**< Amount of encoder subframes. 1 means no subframing. */ + int BSACnumOfSubFrame; /**< The number of the sub-frames which are grouped and transmitted in a super-frame (BSAC). */ + int BSAClayerLength; /**< The average length of the large-step layers in bytes (BSAC). */ + UINT flags; /**< flags */ + UCHAR matrixMixdownA; /**< Matrix mixdown index to put into PCE. Default value 0 means no mixdown coefficient, + valid values are 1-4 which correspond to matrix_mixdown_idx 0-3. */ + UCHAR headerPeriod; /**< Frame period for sending in band configuration buffers in the transport layer. */ + + UCHAR stereoConfigIndex; /**< USAC MPS stereo mode */ + UCHAR sbrMode; /**< USAC SBR mode */ +} CODER_CONFIG; + +/** MP4 Element IDs. */ +typedef enum +{ + ID_NONE = -1, /**< Invalid Element helper ID. */ + ID_SCE = 0, /**< Single Channel Element. */ + ID_CPE = 1, /**< Channel Pair Element. */ + ID_CCE = 2, /**< Coupling Channel Element. */ + ID_LFE = 3, /**< LFE Channel Element. */ + ID_DSE = 4, /**< Currently one Data Stream Element for ancillary data is supported. */ + ID_PCE = 5, /**< Program Config Element. */ + ID_FIL = 6, /**< Fill Element. */ + ID_END = 7, /**< Arnie (End Element = Terminator). */ + ID_EXT = 8, /**< Extension Payload (ER only). */ + ID_SCAL = 9, /**< AAC scalable element (ER only). */ + ID_LAST +} MP4_ELEMENT_ID; + +#define IS_CHANNEL_ELEMENT(elementId) \ + ((elementId) == ID_SCE \ +|| (elementId) == ID_CPE \ +|| (elementId) == ID_LFE) + +#define EXT_ID_BITS 4 /**< Size in bits of extension payload type tags. */ + +/** Extension payload types. */ +typedef enum { + EXT_FIL = 0x00, + EXT_FILL_DATA = 0x01, + EXT_DATA_ELEMENT = 0x02, + EXT_DATA_LENGTH = 0x03, + EXT_LDSAC_DATA = 0x09, + EXT_SAOC_DATA = 0x0a, + EXT_DYNAMIC_RANGE = 0x0b, + EXT_SAC_DATA = 0x0c, + EXT_SBR_DATA = 0x0d, + EXT_SBR_DATA_CRC = 0x0e +} EXT_PAYLOAD_TYPE; + + +/** + * Proprietary raw packet file configuration data type identifier. + */ +typedef enum +{ + TC_NOTHING = 0, /* No configuration available -> in-band configuration. */ + TC_RAW_ASC, /* Configuration data field is a raw AudioSpecificConfig. */ + TC_RAW_SMC, /* Configuration data field is a raw StreamMuxConfig. */ + TC_RAW_SDC /* Configuration data field is a raw Drm SDC. */ + +} TP_CONFIG_TYPE; + +/* + * ############################################################################################## + * Library identification and error handling + * ############################################################################################## + */ +/* \cond */ +#define MODULE_ID_MASK (0x000000ff) +#define MODULE_ID_SHIFT (24) + +typedef enum { + FDK_NONE = 0, + FDK_TOOLS = 1, + FDK_SYSLIB = 2, + FDK_AACDEC = 3, + FDK_AACENC = 4, + FDK_SBRDEC = 5, + FDK_SBRENC = 6, + FDK_TPDEC = 7, + FDK_TPENC = 8, + FDK_MPSDEC = 9, + FDK_MPEGFILEREAD = 10, + FDK_MPEGFILEWRITE = 11, + FDK_MP2DEC = 12, + FDK_DABDEC = 13, + FDK_DABPARSE = 14, + FDK_DRMDEC = 15, + FDK_DRMPARSE = 16, + FDK_AACLDENC = 17, + FDK_MP2ENC = 18, + FDK_MP3ENC = 19, + FDK_MP3DEC = 20, + FDK_MP3HEADPHONE = 21, + FDK_MP3SDEC = 22, + FDK_MP3SENC = 23, + FDK_EAEC = 24, + FDK_DABENC = 25, + FDK_DMBDEC = 26, + FDK_FDREVERB = 27, + FDK_DRMENC = 28, + FDK_METADATATRANSCODER = 29, + FDK_AC3DEC = 30, + FDK_PCMDMX = 31, + + FDK_MODULE_LAST + +} FDK_MODULE_ID; + +/* AAC capability flags */ +#define CAPF_AAC_LC 0x00000001 /**< Support flag for AAC Low Complexity. */ +#define CAPF_ER_AAC_LD 0x00000002 /**< Support flag for AAC Low Delay with Error Resilience tools. */ +#define CAPF_ER_AAC_SCAL 0x00000004 /**< Support flag for AAC Scalable. */ +#define CAPF_ER_AAC_LC 0x00000008 /**< Support flag for AAC Low Complexity with Error Resilience tools. */ +#define CAPF_AAC_480 0x00000010 /**< Support flag for AAC with 480 framelength. */ +#define CAPF_AAC_512 0x00000020 /**< Support flag for AAC with 512 framelength. */ +#define CAPF_AAC_960 0x00000040 /**< Support flag for AAC with 960 framelength. */ +#define CAPF_AAC_1024 0x00000080 /**< Support flag for AAC with 1024 framelength. */ +#define CAPF_AAC_HCR 0x00000100 /**< Support flag for AAC with Huffman Codeword Reordering. */ +#define CAPF_AAC_VCB11 0x00000200 /**< Support flag for AAC Virtual Codebook 11. */ +#define CAPF_AAC_RVLC 0x00000400 /**< Support flag for AAC Reversible Variable Length Coding. */ +#define CAPF_AAC_MPEG4 0x00000800 /**< Support flag for MPEG file format. */ +#define CAPF_AAC_DRC 0x00001000 /**< Support flag for AAC Dynamic Range Control. */ +#define CAPF_AAC_CONCEALMENT 0x00002000 /**< Support flag for AAC concealment. */ +#define CAPF_AAC_DRM_BSFORMAT 0x00004000 /**< Support flag for AAC DRM bistream format. */ +#define CAPF_ER_AAC_ELD 0x00008000 /**< Support flag for AAC Enhanced Low Delay with Error Resilience tools. */ +#define CAPF_ER_AAC_BSAC 0x00010000 /**< Support flag for AAC BSAC. */ +#define CAPF_AAC_SUPERFRAMING 0x00020000 /**< Support flag for AAC Superframing. */ + +/* Transport capability flags */ +#define CAPF_ADTS 0x00000001 /**< Support flag for ADTS transport format. */ +#define CAPF_ADIF 0x00000002 /**< Support flag for ADIF transport format. */ +#define CAPF_LATM 0x00000004 /**< Support flag for LATM transport format. */ +#define CAPF_LOAS 0x00000008 /**< Support flag for LOAS transport format. */ +#define CAPF_RAWPACKETS 0x00000010 /**< Support flag for RAW PACKETS transport format. */ +#define CAPF_DRM 0x00000020 /**< Support flag for DRM/DRM+ transport format. */ +#define CAPF_RSVD50 0x00000040 /**< Support flag for RSVD50 transport format */ + +/* SBR capability flags */ +#define CAPF_SBR_LP 0x00000001 /**< Support flag for SBR Low Power mode. */ +#define CAPF_SBR_HQ 0x00000002 /**< Support flag for SBR High Quality mode. */ +#define CAPF_SBR_DRM_BS 0x00000004 /**< Support flag for */ +#define CAPF_SBR_CONCEALMENT 0x00000008 /**< Support flag for SBR concealment. */ +#define CAPF_SBR_DRC 0x00000010 /**< Support flag for SBR Dynamic Range Control. */ +#define CAPF_SBR_PS_MPEG 0x00000020 /**< Support flag for MPEG Parametric Stereo. */ +#define CAPF_SBR_PS_DRM 0x00000040 /**< Support flag for DRM Parametric Stereo. */ + +/* MP2 encoder capability flags */ +#define CAPF_MP2ENC_SS 0x00000001 /**< Support flag for Seamless Switching. */ +#define CAPF_MP2ENC_DAB 0x00000002 /**< Support flag for Layer2 DAB. */ + +/* DAB capability flags */ +#define CAPF_DAB_MP2 0x00000001 /**< Support flag for Layer2 DAB. */ +#define CAPF_DAB_AAC 0x00000002 /**< Support flag for DAB+ (HE-AAC v2). */ +#define CAPF_DAB_PAD 0x00000004 /**< Support flag for PAD extraction. */ +#define CAPF_DAB_DRC 0x00000008 /**< Support flag for Dynamic Range Control. */ +#define CAPF_DAB_SURROUND 0x00000010 /**< Support flag for DAB Surround (MPS). */ + +/* DMB capability flags */ +#define CAPF_DMB_BSAC 0x00000001 /**< Support flag for ER AAC BSAC. */ +#define CAPF_DMB_DRC 0x00000008 /**< Support flag for Dynamic Range Control. */ +#define CAPF_DMB_SURROUND 0x00000010 /**< Support flag for DMB Surround (MPS). */ + +/* PCM up/downmmix capability flags */ +#define CAPF_DMX_BLIND 0x00000001 /**< Support flag for blind downmixing. */ +#define CAPF_DMX_PCE 0x00000002 /**< Support flag for guided downmix with data from MPEG-2/4 Program Config Elements (PCE). */ +#define CAPF_DMX_ARIB 0x00000004 /**< Support flag for PCE guided downmix with slightly different equations and levels to fulfill ARIB standard. */ +#define CAPF_DMX_DVB 0x00000008 /**< Support flag for guided downmix with data from DVB ancillary data fields. */ +#define CAPF_DMX_CH_EXP 0x00000010 /**< Support flag for simple upmixing by dublicating channels or adding zero channels. */ +/* \endcond */ + + +/* + * ############################################################################################## + * Library versioning + * ############################################################################################## + */ + +/** + * Convert each member of version numbers to one single numeric version representation. + * \param lev0 1st level of version number. + * \param lev1 2nd level of version number. + * \param lev2 3rd level of version number. + */ +#define LIB_VERSION(lev0, lev1, lev2) ((lev0<<24 & 0xff000000) | \ + (lev1<<16 & 0x00ff0000) | \ + (lev2<<8 & 0x0000ff00)) + +/** + * Build text string of version. + */ +#define LIB_VERSION_STRING(info) FDKsprintf((info)->versionStr, "%d.%d.%d", (((info)->version >> 24) & 0xff), (((info)->version >> 16) & 0xff), (((info)->version >> 8 ) & 0xff)) + +/** + * Library information. + */ +typedef struct LIB_INFO +{ + const char* title; + const char* build_date; + const char* build_time; + FDK_MODULE_ID module_id; + INT version; + UINT flags; + char versionStr[32]; +} LIB_INFO; + +/** Initialize library info. */ +static inline void FDKinitLibInfo( LIB_INFO* info ) +{ + int i; + + for (i = 0; i < FDK_MODULE_LAST; i++) { + info[i].module_id = FDK_NONE; + } +} + +/** Aquire supported features of library. */ +static inline UINT FDKlibInfo_getCapabilities( const LIB_INFO* info, FDK_MODULE_ID module_id ) +{ + int i; + + for (i=0; i<FDK_MODULE_LAST; i++) { + if (info[i].module_id == module_id) { + return info[i].flags; + } + } + return 0; +} + +/** Search for next free tab. */ +static inline INT FDKlibInfo_lookup( const LIB_INFO* info, FDK_MODULE_ID module_id ) +{ + int i = -1; + + for (i = 0; i < FDK_MODULE_LAST; i++) { + if (info[i].module_id == module_id) + return -1; + if (info[i].module_id == FDK_NONE) + break; + } + if (i == FDK_MODULE_LAST) + return -1; + + return i; +} + + +/* + * ############################################################################################## + * Buffer description + * ############################################################################################## + */ + +/** + * I/O buffer descriptor. + */ +typedef struct FDK_bufDescr +{ + void **ppBase; /*!< Pointer to an array containing buffer base addresses. + Set to NULL for buffer requirement info. */ + UINT *pBufSize; /*!< Pointer to an array containing the number of elements that can + be placed in the specific buffer. */ + UINT *pEleSize; /*!< Pointer to an array containing the element size for each buffer + in bytes. That is mostly the number returned by the sizeof() + operator for the data type used for the specific buffer. */ + UINT *pBufType; /*!< Pointer to an array of bit fields containing a description + for each buffer. See XXX below for more details. */ + UINT numBufs; /*!< Total number of buffers. */ + +} FDK_bufDescr; + +/** + * Buffer type description field. + */ +#define FDK_BUF_TYPE_MASK_IO ( 0x03 << 30 ) +#define FDK_BUF_TYPE_MASK_DESCR ( 0x3F << 16 ) +#define FDK_BUF_TYPE_MASK_ID ( 0xFF ) + +#define FDK_BUF_TYPE_INPUT ( 0x1 << 30 ) +#define FDK_BUF_TYPE_OUTPUT ( 0x2 << 30 ) + +#define FDK_BUF_TYPE_PCM_DATA ( 0x1 << 16 ) +#define FDK_BUF_TYPE_ANC_DATA ( 0x2 << 16 ) +#define FDK_BUF_TYPE_BS_DATA ( 0x4 << 16 ) + +#ifdef __cplusplus +} +#endif + +#endif /* FDK_AUDIO_H */ |