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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 10:03:58 +0200 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 10:03:58 +0200 |
commit | a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d (patch) | |
tree | 2b4790eec8f47fb086e645717f07c53b30ace919 /fdk-aac/libSYS/include/FDK_audio.h | |
parent | 2f84a54ec1d10b10293c7b1f4ab9fee31f3c6327 (diff) | |
parent | c6a73c219dbfdfe639372d9922f4eb512f06fa2f (diff) | |
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Merge GStreamer into next
Diffstat (limited to 'fdk-aac/libSYS/include/FDK_audio.h')
-rw-r--r-- | fdk-aac/libSYS/include/FDK_audio.h | 827 |
1 files changed, 827 insertions, 0 deletions
diff --git a/fdk-aac/libSYS/include/FDK_audio.h b/fdk-aac/libSYS/include/FDK_audio.h new file mode 100644 index 0000000..d69c008 --- /dev/null +++ b/fdk-aac/libSYS/include/FDK_audio.h @@ -0,0 +1,827 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/************************* System integration library ************************** + + Author(s): Manuel Jander + + Description: + +*******************************************************************************/ + +/** \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" +#include "syslib_channelMapDescr.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. */ + +} 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_DABPLUS = 13 /**< Digital Audio Broadcastong (DAB+) superframes bitstream format. */ + +} TRANSPORT_TYPE; + +#define TT_IS_PACKET(x) \ + (((x) == TT_MP4_RAW) || ((x) == TT_DRM) || ((x) == TT_MP4_LATM_MCP0) || \ + ((x) == TT_MP4_LATM_MCP1)) + +/** + * 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_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 */ + + + /* Pseudo AOTs */ + AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */ + AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */ + + 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_DRM_USAC = 147 /**< Virtual AOT for DRM with USAC */ + +} AUDIO_OBJECT_TYPE; + +#define CAN_DO_PS(aot) \ + ((aot) == AOT_AAC_LC || (aot) == AOT_SBR || (aot) == AOT_PS || \ + (aot) == AOT_ER_BSAC || (aot) == AOT_DRM_AAC) + +#define IS_USAC(aot) ((aot) == AOT_USAC) + +#define IS_LOWDELAY(aot) ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD) + +/** Channel Mode ( 1-7 equals MPEG channel configurations, others are + * arbitrary). */ +typedef enum { + MODE_INVALID = -1, + MODE_UNKNOWN = 0, + MODE_1 = 1, /**< C */ + MODE_2 = 2, /**< L+R */ + MODE_1_2 = 3, /**< C, L+R */ + MODE_1_2_1 = 4, /**< C, L+R, Rear */ + MODE_1_2_2 = 5, /**< C, L+R, LS+RS */ + MODE_1_2_2_1 = 6, /**< C, L+R, LS+RS, LFE */ + MODE_1_2_2_2_1 = 7, /**< C, LC+RC, L+R, LS+RS, LFE */ + + MODE_6_1 = 11, /**< C, L+R, LS+RS, Crear, LFE */ + MODE_7_1_BACK = 12, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */ + MODE_7_1_TOP_FRONT = 14, /**< C, L+R, LS+RS, LFE, Ltop+Rtop */ + + MODE_7_1_REAR_SURROUND = 33, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */ + MODE_7_1_FRONT_CENTER = 34, /**< C, LC+RC, L+R, LS+RS, LFE */ + + MODE_212 = 128 /**< 212 configuration, used in ELDv2 */ + +} CHANNEL_MODE; + +/** + * Speaker description tags. + * Do not change the enumeration values unless it keeps the following + * segmentation: + * - Bit 0-3: Horizontal postion (0: none, 1: front, 2: side, 3: back, 4: lfe) + * - Bit 4-7: Vertical position (0: normal, 1: top, 2: bottom) + */ +typedef enum { + ACT_NONE = 0x00, + ACT_FRONT = 0x01, /*!< Front speaker position (at normal height) */ + ACT_SIDE = 0x02, /*!< Side speaker position (at normal height) */ + ACT_BACK = 0x03, /*!< Back speaker position (at normal height) */ + ACT_LFE = 0x04, /*!< Low frequency effect speaker postion (front) */ + + ACT_TOP = + 0x10, /*!< Top speaker area (for combination with speaker positions) */ + ACT_FRONT_TOP = 0x11, /*!< Top front speaker = (ACT_FRONT|ACT_TOP) */ + ACT_SIDE_TOP = 0x12, /*!< Top side speaker = (ACT_SIDE |ACT_TOP) */ + ACT_BACK_TOP = 0x13, /*!< Top back speaker = (ACT_BACK |ACT_TOP) */ + + ACT_BOTTOM = + 0x20, /*!< Bottom speaker area (for combination with speaker positions) */ + ACT_FRONT_BOTTOM = 0x21, /*!< Bottom front speaker = (ACT_FRONT|ACT_BOTTOM) */ + ACT_SIDE_BOTTOM = 0x22, /*!< Bottom side speaker = (ACT_SIDE |ACT_BOTTOM) */ + ACT_BACK_BOTTOM = 0x23 /*!< Bottom back speaker = (ACT_BACK |ACT_BOTTOM) */ + +} AUDIO_CHANNEL_TYPE; + +typedef enum { + SIG_UNKNOWN = -1, + SIG_IMPLICIT = 0, + SIG_EXPLICIT_BW_COMPATIBLE = 1, + SIG_EXPLICIT_HIERARCHICAL = 2 + +} SBR_PS_SIGNALING; + +/** + * 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_RSV603DA 0x000200 /*!< RSVD60 3D audio */ +#define AC_HDAAC 0x000400 /*!< HD-AAC */ +#define AC_RSVD50 0x004000 /*!< Rsvd50 */ +#define AC_SBR_PRESENT 0x008000 /*!< SBR present flag (from ASC) */ +#define AC_SBRCRC \ + 0x010000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */ +#define AC_PS_PRESENT 0x020000 /*!< PS present flag (from ASC or implicit) */ +#define AC_MPS_PRESENT \ + 0x040000 /*!< MPS present flag (from ASC or implicit) \ + */ +#define AC_DRM 0x080000 /*!< DRM bit stream syntax */ +#define AC_INDEP 0x100000 /*!< Independency flag */ +#define AC_MPEGD_RES 0x200000 /*!< MPEG-D residual individual channel data. */ +#define AC_SAOC_PRESENT 0x400000 /*!< SAOC Present Flag */ +#define AC_DAB 0x800000 /*!< DAB bit stream syntax */ +#define AC_ELD_DOWNSCALE 0x1000000 /*!< ELD Downscaled playout */ +#define AC_LD_MPS 0x2000000 /*!< Low Delay MPS. */ +#define AC_DRC_PRESENT \ + 0x4000000 /*!< Dynamic Range Control (DRC) data found. \ + */ +#define AC_USAC_SCFGI3 \ + 0x8000000 /*!< USAC flag: If stereoConfigIndex is 3 the flag is set. */ +/** + * Audio Codec flags (reconfiguration). + */ +#define AC_CM_DET_CFG_CHANGE \ + 0x000001 /*!< Config mode signalizes the callback to work in config change \ + detection mode */ +#define AC_CM_ALLOC_MEM \ + 0x000002 /*!< Config mode signalizes the callback to work in memory \ + allocation mode */ + +/** + * Audio Codec flags (element specific). + */ +#define AC_EL_USAC_TW 0x000001 /*!< USAC time warped filter bank is active */ +#define AC_EL_USAC_NOISE 0x000002 /*!< USAC noise filling is active */ +#define AC_EL_USAC_ITES 0x000004 /*!< USAC SBR inter-TES tool is active */ +#define AC_EL_USAC_PVC \ + 0x000008 /*!< USAC SBR predictive vector coding tool is active */ +#define AC_EL_USAC_MPS212 0x000010 /*!< USAC MPS212 tool is active */ +#define AC_EL_USAC_LFE 0x000020 /*!< USAC element is LFE */ +#define AC_EL_USAC_CP_POSSIBLE \ + 0x000040 /*!< USAC may use Complex Stereo Prediction in this channel element \ + */ +#define AC_EL_ENHANCED_NOISE 0x000080 /*!< Enhanced noise filling*/ +#define AC_EL_IGF_AFTER_TNS 0x000100 /*!< IGF after TNS */ +#define AC_EL_IGF_INDEP_TILING 0x000200 /*!< IGF independent tiling */ +#define AC_EL_IGF_USE_ENF 0x000400 /*!< IGF use enhanced noise filling */ +#define AC_EL_FULLBANDLPD 0x000800 /*!< enable fullband LPD tools */ +#define AC_EL_LPDSTEREOIDX 0x001000 /*!< LPD-stereo-tool stereo index */ +#define AC_EL_LFE 0x002000 /*!< The element is of type LFE. */ + +/* 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_SAC 0x00020000 +#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. */ + UCHAR channelConfigZero; /**< Use channel config zero + pce although a + standard channel config could be signaled. */ + INT samplingRate; /**< Sampling rate. */ + INT extSamplingRate; /**< Extended samplerate (SBR). */ + INT downscaleSamplingRate; /**< Downscale sampling rate (ELD downscaled mode) + */ + 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 */ + SBR_PS_SIGNALING sbrSignaling; /**< 0: implicit signaling, 1: backwards + compatible explicit signaling, 2: + hierarcical explicit signaling */ + + UCHAR rawConfig[64]; /**< raw codec specific config as bit stream */ + int rawConfigBits; /**< Size of rawConfig in bits */ + + UCHAR sbrPresent; + UCHAR psPresent; +} CODER_CONFIG; + +#define USAC_ID_BIT 16 /** USAC element IDs start at USAC_ID_BIT */ + +/** MP4 Element IDs. */ +typedef enum { + /* mp4 element IDs */ + 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). */ + /* USAC element IDs */ + ID_USAC_SCE = 0 + USAC_ID_BIT, /**< Single Channel Element. */ + ID_USAC_CPE = 1 + USAC_ID_BIT, /**< Channel Pair Element. */ + ID_USAC_LFE = 2 + USAC_ID_BIT, /**< LFE Channel Element. */ + ID_USAC_EXT = 3 + USAC_ID_BIT, /**< Extension Element. */ + ID_USAC_END = 4 + USAC_ID_BIT, /**< Arnie (End Element = Terminator). */ + ID_LAST +} MP4_ELEMENT_ID; + +/* usacConfigExtType q.v. ISO/IEC DIS 23008-3 Table 52 and ISO/IEC FDIS + * 23003-3:2011(E) Table 74*/ +typedef enum { + /* USAC and RSVD60 3DA */ + ID_CONFIG_EXT_FILL = 0, + /* RSVD60 3DA */ + ID_CONFIG_EXT_DOWNMIX = 1, + ID_CONFIG_EXT_LOUDNESS_INFO = 2, + ID_CONFIG_EXT_AUDIOSCENE_INFO = 3, + ID_CONFIG_EXT_HOA_MATRIX = 4, + ID_CONFIG_EXT_SIG_GROUP_INFO = 6 + /* 5-127 => reserved for ISO use */ + /* > 128 => reserved for use outside of ISO scope */ +} CONFIG_EXT_ID; + +#define IS_CHANNEL_ELEMENT(elementId) \ + ((elementId) == ID_SCE || (elementId) == ID_CPE || (elementId) == ID_LFE || \ + (elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE || \ + (elementId) == ID_USAC_LFE) + +#define IS_MP4_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_UNI_DRC = 0x04, + 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; + +#define IS_USAC_CHANNEL_ELEMENT(elementId) \ + ((elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE || \ + (elementId) == ID_USAC_LFE) + +/** MPEG-D USAC & RSVD60 3D audio Extension Element Types. */ +typedef enum { + /* usac */ + ID_EXT_ELE_FILL = 0x00, + ID_EXT_ELE_MPEGS = 0x01, + ID_EXT_ELE_SAOC = 0x02, + ID_EXT_ELE_AUDIOPREROLL = 0x03, + ID_EXT_ELE_UNI_DRC = 0x04, + /* rsv603da */ + ID_EXT_ELE_OBJ_METADATA = 0x05, + ID_EXT_ELE_SAOC_3D = 0x06, + ID_EXT_ELE_HOA = 0x07, + ID_EXT_ELE_FMT_CNVRTR = 0x08, + ID_EXT_ELE_MCT = 0x09, + ID_EXT_ELE_ENHANCED_OBJ_METADATA = 0x0d, + /* reserved for use outside of ISO scope */ + ID_EXT_ELE_VR_METADATA = 0x81, + ID_EXT_ELE_UNKNOWN = 0xFF +} USAC_EXT_ELEMENT_TYPE; + +/** + * Proprietary raw packet file configuration data type identifier. + */ +typedef enum { + TC_NOTHING = 0, /* No configuration available -> in-band configuration. */ + TC_RAW_ADTS = 2, /* Transfer type is ADTS. */ + TC_RAW_LATM_MCP1 = 6, /* Transfer type is LATM with SMC present. */ + TC_RAW_SDC = 21 /* Configuration data field is Drm SDC. */ + +} TP_CONFIG_TYPE; + +/* + * ############################################################################################## + * Library identification and error handling + * ############################################################################################## + */ +/* \cond */ + +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_PCMDMX = 31, + FDK_MPSENC = 34, + FDK_TDLIMIT = 35, + FDK_UNIDRCDEC = 38, + + 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_ELD_DOWNSCALE \ + 0x00040000 /**< Support flag for AAC-ELD Downscaling */ +#define CAPF_AAC_USAC_LP \ + 0x00100000 /**< Support flag for USAC low power mode. */ +#define CAPF_AAC_USAC \ + 0x00200000 /**< Support flag for Unified Speech and Audio Coding (USAC). */ +#define CAPF_ER_AAC_ELDV2 \ + 0x00800000 /**< Support flag for AAC Enhanced Low Delay with MPS 212. */ +#define CAPF_AAC_UNIDRC \ + 0x01000000 /**< Support flag for MPEG-D Dynamic Range Control (uniDrc). */ + +/* 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. */ +#define CAPF_SBR_ELD_DOWNSCALE \ + 0x00000080 /**< Support flag for ELD reduced delay mode */ +#define CAPF_SBR_HBEHQ \ + 0x00000100 /**< Support flag for HQ HBE */ + +/* 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). */ + + +/* PCM utils 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. */ +#define CAPF_DMX_6_CH \ + 0x00000020 /**< Support flag for 5.1 channel configuration (input and \ + output). */ +#define CAPF_DMX_8_CH \ + 0x00000040 /**< Support flag for 6 and 7.1 channel configurations (input and \ + output). */ +#define CAPF_DMX_24_CH \ + 0x00000080 /**< Support flag for 22.2 channel configuration (input and \ + output). */ +#define CAPF_LIMITER \ + 0x00002000 /**< Support flag for signal level limiting. \ + */ + +/* MPEG Surround capability flags */ +#define CAPF_MPS_STD \ + 0x00000001 /**< Support flag for MPEG Surround. */ +#define CAPF_MPS_LD \ + 0x00000002 /**< Support flag for Low Delay MPEG Surround. \ + */ +#define CAPF_MPS_USAC \ + 0x00000004 /**< Support flag for USAC MPEG Surround. */ +#define CAPF_MPS_HQ \ + 0x00000010 /**< Support flag indicating if high quality processing is \ + supported */ +#define CAPF_MPS_LP \ + 0x00000020 /**< Support flag indicating if partially complex (low power) \ + processing is supported */ +#define CAPF_MPS_BLIND \ + 0x00000040 /**< Support flag indicating if blind processing is supported */ +#define CAPF_MPS_BINAURAL \ + 0x00000080 /**< Support flag indicating if binaural output is possible */ +#define CAPF_MPS_2CH_OUT \ + 0x00000100 /**< Support flag indicating if 2ch output is possible */ +#define CAPF_MPS_6CH_OUT \ + 0x00000200 /**< Support flag indicating if 6ch output is possible */ +#define CAPF_MPS_8CH_OUT \ + 0x00000400 /**< Support flag indicating if 8ch output is possible */ +#define CAPF_MPS_1CH_IN \ + 0x00001000 /**< Support flag indicating if 1ch dmx input is possible */ +#define CAPF_MPS_2CH_IN \ + 0x00002000 /**< Support flag indicating if 2ch dmx input is possible */ +#define CAPF_MPS_6CH_IN \ + 0x00004000 /**< Support flag indicating if 5ch dmx input is possible */ + +/* \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; + +#ifdef __cplusplus +#define FDK_AUDIO_INLINE inline +#else +#define FDK_AUDIO_INLINE +#endif + +/** Initialize library info. */ +static FDK_AUDIO_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 FDK_AUDIO_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 FDK_AUDIO_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 ((UINT)0x03 << 30) +#define FDK_BUF_TYPE_MASK_DESCR ((UINT)0x3F << 16) +#define FDK_BUF_TYPE_MASK_ID ((UINT)0xFF) + +#define FDK_BUF_TYPE_INPUT ((UINT)0x1 << 30) +#define FDK_BUF_TYPE_OUTPUT ((UINT)0x2 << 30) + +#define FDK_BUF_TYPE_PCM_DATA ((UINT)0x1 << 16) +#define FDK_BUF_TYPE_ANC_DATA ((UINT)0x2 << 16) +#define FDK_BUF_TYPE_BS_DATA ((UINT)0x4 << 16) + +#ifdef __cplusplus +} +#endif + +#endif /* FDK_AUDIO_H */ |