mirror of https://github.com/mstorsjo/fdk-aac.git
613 lines
27 KiB
C
613 lines
27 KiB
C
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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/************************** Fraunhofer IIS FDK SysLib **********************
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Author(s): Manuel Jander
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******************************************************************************/
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/** \file FDK_audio.h
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* \brief Global audio struct and constant definitions.
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*/
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#ifndef FDK_AUDIO_H
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#define FDK_AUDIO_H
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#include "machine_type.h"
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#include "genericStds.h"
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#ifdef __cplusplus
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extern "C"
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{
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#endif
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/**
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* File format identifiers.
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*/
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typedef enum
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{
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FF_UNKNOWN = -1, /**< Unknown format. */
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FF_RAW = 0, /**< No container, bit stream data conveyed "as is". */
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FF_MP4_3GPP = 3, /**< 3GPP file format. */
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FF_MP4_MP4F = 4, /**< MPEG-4 File format. */
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FF_RAWPACKETS = 5, /**< Proprietary raw packet file. */
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FF_DRMCT = 12 /**< Digital Radio Mondial (DRM30/DRM+) CT proprietary file format. */
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} FILE_FORMAT;
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/**
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* Transport type identifiers.
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*/
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typedef enum
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{
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TT_UNKNOWN = -1, /**< Unknown format. */
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TT_MP4_RAW = 0, /**< "as is" access units (packet based since there is obviously no sync layer) */
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TT_MP4_ADIF = 1, /**< ADIF bitstream format. */
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TT_MP4_ADTS = 2, /**< ADTS bitstream format. */
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TT_MP4_LATM_MCP1 = 6, /**< Audio Mux Elements with muxConfigPresent = 1 */
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TT_MP4_LATM_MCP0 = 7, /**< Audio Mux Elements with muxConfigPresent = 0, out of band StreamMuxConfig */
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TT_MP4_LOAS = 10, /**< Audio Sync Stream. */
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TT_DRM = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
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TT_MP1_L1 = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */
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TT_MP1_L2 = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */
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TT_MP1_L3 = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */
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TT_RSVD50 = 50 /**< */
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} TRANSPORT_TYPE;
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/**
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* Audio Object Type definitions.
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*/
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typedef enum
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{
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AOT_NONE = -1,
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AOT_NULL_OBJECT = 0,
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AOT_AAC_MAIN = 1, /**< Main profile */
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AOT_AAC_LC = 2, /**< Low Complexity object */
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AOT_AAC_SSR = 3,
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AOT_AAC_LTP = 4,
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AOT_SBR = 5,
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AOT_AAC_SCAL = 6,
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AOT_TWIN_VQ = 7,
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AOT_CELP = 8,
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AOT_HVXC = 9,
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AOT_RSVD_10 = 10, /**< (reserved) */
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AOT_RSVD_11 = 11, /**< (reserved) */
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AOT_TTSI = 12, /**< TTSI Object */
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AOT_MAIN_SYNTH = 13, /**< Main Synthetic object */
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AOT_WAV_TAB_SYNTH = 14, /**< Wavetable Synthesis object */
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AOT_GEN_MIDI = 15, /**< General MIDI object */
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AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
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AOT_ER_AAC_LC = 17, /**< Error Resilient(ER) AAC Low Complexity */
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AOT_RSVD_18 = 18, /**< (reserved) */
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AOT_ER_AAC_LTP = 19, /**< Error Resilient(ER) AAC LTP object */
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AOT_ER_AAC_SCAL = 20, /**< Error Resilient(ER) AAC Scalable object */
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AOT_ER_TWIN_VQ = 21, /**< Error Resilient(ER) TwinVQ object */
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AOT_ER_BSAC = 22, /**< Error Resilient(ER) BSAC object */
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AOT_ER_AAC_LD = 23, /**< Error Resilient(ER) AAC LowDelay object */
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AOT_ER_CELP = 24, /**< Error Resilient(ER) CELP object */
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AOT_ER_HVXC = 25, /**< Error Resilient(ER) HVXC object */
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AOT_ER_HILN = 26, /**< Error Resilient(ER) HILN object */
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AOT_ER_PARA = 27, /**< Error Resilient(ER) Parametric object */
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AOT_RSVD_28 = 28, /**< might become SSC */
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AOT_PS = 29, /**< PS, Parametric Stereo (includes SBR) */
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AOT_MPEGS = 30, /**< MPEG Surround */
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AOT_ESCAPE = 31, /**< Signal AOT uses more than 5 bits */
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AOT_MP3ONMP4_L1 = 32, /**< MPEG-Layer1 in mp4 */
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AOT_MP3ONMP4_L2 = 33, /**< MPEG-Layer2 in mp4 */
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AOT_MP3ONMP4_L3 = 34, /**< MPEG-Layer3 in mp4 */
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AOT_RSVD_35 = 35, /**< might become DST */
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AOT_RSVD_36 = 36, /**< might become ALS */
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AOT_AAC_SLS = 37, /**< AAC + SLS */
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AOT_SLS = 38, /**< SLS */
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AOT_ER_AAC_ELD = 39, /**< AAC Enhanced Low Delay */
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AOT_USAC = 42, /**< USAC */
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AOT_SAOC = 43, /**< SAOC */
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AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround */
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AOT_RSVD50 = 50, /**< Interim AOT for Rsvd50 */
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/* Pseudo AOTs */
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AOT_MP2_AAC_MAIN = 128, /**< Virtual AOT MP2 Main profile */
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AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */
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AOT_MP2_AAC_SSR = 130, /**< Virtual AOT MP2 Scalable Sampling Rate profile */
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AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */
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AOT_DAB = 134, /**< Virtual AOT for DAB (Layer2 with scalefactor CRC) */
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AOT_DABPLUS_AAC_LC = 135, /**< Virtual AOT for DAB plus AAC-LC */
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AOT_DABPLUS_SBR = 136, /**< Virtual AOT for DAB plus HE-AAC */
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AOT_DABPLUS_PS = 137, /**< Virtual AOT for DAB plus HE-AAC v2 */
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AOT_PLAIN_MP1 = 140, /**< Virtual AOT for plain mp1 */
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AOT_PLAIN_MP2 = 141, /**< Virtual AOT for plain mp2 */
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AOT_PLAIN_MP3 = 142, /**< Virtual AOT for plain mp3 */
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AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */
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AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */
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AOT_DRM_MPEG_PS = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
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AOT_DRM_SURROUND = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
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AOT_MP2_PS = 156, /**< Virtual AOT MP2 Low Complexity Profile with SBR and PS */
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AOT_MPEGS_RESIDUALS = 256 /**< Virtual AOT for MPEG Surround residuals */
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} AUDIO_OBJECT_TYPE;
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/** Channel Mode ( 1-7 equals MPEG channel configurations, others are arbitrary). */
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typedef enum {
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MODE_INVALID = -1,
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MODE_UNKNOWN = 0,
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MODE_1 = 1, /**< SCE */
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MODE_2 = 2, /**< CPE */
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MODE_1_2 = 3, /**< SCE,CPE */
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MODE_1_2_1 = 4, /**< SCE,CPE,SCE */
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MODE_1_2_2 = 5, /**< SCE,CPE,CPE */
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MODE_1_2_2_1 = 6, /**< SCE,CPE,CPE,LFE */
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MODE_1_2_2_2_1 = 7, /**< SCE,CPE,CPE,CPE,LFE */
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MODE_1_1 = 16, /**< 2 SCEs (dual mono) */
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MODE_1_1_1_1 = 17, /**< 4 SCEs */
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MODE_1_1_1_1_1_1 = 18, /**< 6 SCEs */
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MODE_1_1_1_1_1_1_1_1 = 19, /**< 8 SCEs */
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MODE_1_1_1_1_1_1_1_1_1_1_1_1 = 20, /**< 12 SCEs */
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MODE_2_2 = 21, /**< 2 CPEs */
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MODE_2_2_2 = 22, /**< 3 CPEs */
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MODE_2_2_2_2 = 23, /**< 4 CPEs */
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MODE_2_2_2_2_2_2 = 24, /**< 6 CPEs */
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MODE_2_1 = 30 /**< CPE,SCE (ARIB standard) */
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} CHANNEL_MODE;
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/** Speaker description tags */
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typedef enum {
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ACT_NONE,
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ACT_FRONT,
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ACT_SIDE,
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ACT_BACK,
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ACT_LFE,
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ACT_FRONT_TOP,
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ACT_SIDE_TOP,
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ACT_BACK_TOP,
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ACT_TOP /* Ts */
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} AUDIO_CHANNEL_TYPE;
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/**
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* Audio Codec flags.
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*/
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#define AC_ER_VCB11 0x000001 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use virtual codebooks */
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#define AC_ER_RVLC 0x000002 /*!< aacSpectralDataResilienceFlag flag (from ASC): 1 means use huffman codeword reordering */
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#define AC_ER_HCR 0x000004 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use virtual codebooks */
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#define AC_SCALABLE 0x000008 /*!< AAC Scalable*/
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#define AC_ELD 0x000010 /*!< AAC-ELD */
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#define AC_LD 0x000020 /*!< AAC-LD */
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#define AC_ER 0x000040 /*!< ER syntax */
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#define AC_BSAC 0x000080 /*!< BSAC */
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#define AC_USAC 0x000100 /*!< USAC */
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#define AC_USAC_TW 0x000200 /*!< USAC time warped filter bank is active */
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#define AC_USAC_NOISE 0x000400 /*!< USAC noise filling is active */
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#define AC_USAC_HBE 0x000800 /*!< USAC harmonic bandwidth extension is active */
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#define AC_RSVD50 0x001000 /*!< Rsvd50 */
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#define AC_SBR_PRESENT 0x002000 /*!< SBR present flag (from ASC) */
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#define AC_SBRCRC 0x004000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */
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#define AC_PS_PRESENT 0x008000 /*!< PS present flag (from ASC or implicit) */
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#define AC_MPS_PRESENT 0x010000 /*!< MPS present flag (from ASC or implicit) */
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#define AC_DRM 0x020000 /*!< DRM bit stream syntax */
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#define AC_INDEP 0x040000 /*!< Independency flag */
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#define AC_MPS_RES 0x080000 /*!< MPS residual individual channel data. */
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#define AC_DAB 0x800000 /*!< DAB bit stream syntax */
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#define AC_LD_MPS 0x01000000 /*!< Low Delay MPS. */
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/* CODER_CONFIG::flags */
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#define CC_MPEG_ID 0x00100000
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#define CC_IS_BASELAYER 0x00200000
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#define CC_PROTECTION 0x00400000
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#define CC_SBR 0x00800000
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#define CC_SBRCRC 0x00010000
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#define CC_RVLC 0x01000000
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#define CC_VCB11 0x02000000
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#define CC_HCR 0x04000000
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#define CC_PSEUDO_SURROUND 0x08000000
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#define CC_USAC_NOISE 0x10000000
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#define CC_USAC_TW 0x20000000
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#define CC_USAC_HBE 0x40000000
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/** Generic audio coder configuration structure. */
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typedef struct {
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AUDIO_OBJECT_TYPE aot; /**< Audio Object Type (AOT). */
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AUDIO_OBJECT_TYPE extAOT; /**< Extension Audio Object Type (SBR). */
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CHANNEL_MODE channelMode; /**< Channel mode. */
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INT samplingRate; /**< Sampling rate. */
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INT extSamplingRate; /**< Extended samplerate (SBR). */
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INT bitRate; /**< Average bitrate. */
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int samplesPerFrame; /**< Number of PCM samples per codec frame and audio channel. */
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int noChannels; /**< Number of audio channels. */
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int bitsFrame;
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int nSubFrames; /**< Amount of encoder subframes. 1 means no subframing. */
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int BSACnumOfSubFrame; /**< The number of the sub-frames which are grouped and transmitted in a super-frame (BSAC). */
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int BSAClayerLength; /**< The average length of the large-step layers in bytes (BSAC). */
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UINT flags; /**< flags */
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UCHAR matrixMixdownA; /**< Matrix mixdown index to put into PCE. Default value 0 means no mixdown coefficient,
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valid values are 1-4 which correspond to matrix_mixdown_idx 0-3. */
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UCHAR headerPeriod; /**< Frame period for sending in band configuration buffers in the transport layer. */
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UCHAR stereoConfigIndex; /**< USAC MPS stereo mode */
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UCHAR sbrMode; /**< USAC SBR mode */
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} CODER_CONFIG;
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/** MP4 Element IDs. */
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typedef enum
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{
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ID_NONE = -1, /**< Invalid Element helper ID. */
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ID_SCE = 0, /**< Single Channel Element. */
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ID_CPE = 1, /**< Channel Pair Element. */
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ID_CCE = 2, /**< Coupling Channel Element. */
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ID_LFE = 3, /**< LFE Channel Element. */
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ID_DSE = 4, /**< Currently one Data Stream Element for ancillary data is supported. */
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ID_PCE = 5, /**< Program Config Element. */
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ID_FIL = 6, /**< Fill Element. */
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ID_END = 7, /**< Arnie (End Element = Terminator). */
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ID_EXT = 8, /**< Extension Payload (ER only). */
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ID_SCAL = 9, /**< AAC scalable element (ER only). */
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ID_LAST
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} MP4_ELEMENT_ID;
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#define IS_CHANNEL_ELEMENT(elementId) \
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((elementId) == ID_SCE \
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|| (elementId) == ID_CPE \
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|| (elementId) == ID_LFE)
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#define EXT_ID_BITS 4 /**< Size in bits of extension payload type tags. */
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/** Extension payload types. */
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typedef enum {
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EXT_FIL = 0x00,
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EXT_FILL_DATA = 0x01,
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EXT_DATA_ELEMENT = 0x02,
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EXT_DATA_LENGTH = 0x03,
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EXT_LDSAC_DATA = 0x09,
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EXT_SAOC_DATA = 0x0a,
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EXT_DYNAMIC_RANGE = 0x0b,
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EXT_SAC_DATA = 0x0c,
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EXT_SBR_DATA = 0x0d,
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EXT_SBR_DATA_CRC = 0x0e
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} EXT_PAYLOAD_TYPE;
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/**
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* Proprietary raw packet file configuration data type identifier.
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*/
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typedef enum
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{
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TC_NOTHING = 0, /* No configuration available -> in-band configuration. */
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TC_RAW_ASC, /* Configuration data field is a raw AudioSpecificConfig. */
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TC_RAW_SMC, /* Configuration data field is a raw StreamMuxConfig. */
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TC_RAW_SDC /* Configuration data field is a raw Drm SDC. */
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} TP_CONFIG_TYPE;
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/*
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* ##############################################################################################
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* Library identification and error handling
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* ##############################################################################################
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*/
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/* \cond */
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#define MODULE_ID_MASK (0x000000ff)
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#define MODULE_ID_SHIFT (24)
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typedef enum {
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FDK_NONE = 0,
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FDK_TOOLS = 1,
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FDK_SYSLIB = 2,
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FDK_AACDEC = 3,
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FDK_AACENC = 4,
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FDK_SBRDEC = 5,
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FDK_SBRENC = 6,
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FDK_TPDEC = 7,
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FDK_TPENC = 8,
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FDK_MPSDEC = 9,
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FDK_MPEGFILEREAD = 10,
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FDK_MPEGFILEWRITE = 11,
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FDK_MP2DEC = 12,
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FDK_DABDEC = 13,
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FDK_DABPARSE = 14,
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FDK_DRMDEC = 15,
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FDK_DRMPARSE = 16,
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FDK_AACLDENC = 17,
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FDK_MP2ENC = 18,
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FDK_MP3ENC = 19,
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FDK_MP3DEC = 20,
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FDK_MP3HEADPHONE = 21,
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FDK_MP3SDEC = 22,
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FDK_MP3SENC = 23,
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FDK_EAEC = 24,
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FDK_DABENC = 25,
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FDK_DMBDEC = 26,
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FDK_FDREVERB = 27,
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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 */
|