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814 lines
33 KiB
814 lines
33 KiB
/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. 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
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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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
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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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,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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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
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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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
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized 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
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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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,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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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
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this software, even if 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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/************************* System integration library **************************
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Author(s): Manuel Jander
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Description:
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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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#include "syslib_channelMapDescr.h"
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#ifdef __cplusplus
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extern "C" {
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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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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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} 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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TT_UNKNOWN = -1, /**< Unknown format. */
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TT_MP4_RAW = 0, /**< "as is" access units (packet based since there is
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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
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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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} TRANSPORT_TYPE;
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#define TT_IS_PACKET(x) \
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(((x) == TT_MP4_RAW) || ((x) == TT_DRM) || ((x) == TT_MP4_LATM_MCP0) || \
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((x) == TT_MP4_LATM_MCP1))
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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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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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/* Pseudo AOTs */
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AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */
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AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR */
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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 =
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145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
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AOT_DRM_SURROUND =
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146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
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AOT_DRM_USAC = 147 /**< Virtual AOT for DRM with USAC */
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} AUDIO_OBJECT_TYPE;
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#define CAN_DO_PS(aot) \
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((aot) == AOT_AAC_LC || (aot) == AOT_SBR || (aot) == AOT_PS || \
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(aot) == AOT_ER_BSAC || (aot) == AOT_DRM_AAC)
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#define IS_USAC(aot) ((aot) == AOT_USAC)
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#define IS_LOWDELAY(aot) ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD)
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/** Channel Mode ( 1-7 equals MPEG channel configurations, others are
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* 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, /**< C */
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MODE_2 = 2, /**< L+R */
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MODE_1_2 = 3, /**< C, L+R */
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MODE_1_2_1 = 4, /**< C, L+R, Rear */
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MODE_1_2_2 = 5, /**< C, L+R, LS+RS */
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MODE_1_2_2_1 = 6, /**< C, L+R, LS+RS, LFE */
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MODE_1_2_2_2_1 = 7, /**< C, LC+RC, L+R, LS+RS, LFE */
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MODE_6_1 = 11, /**< C, L+R, LS+RS, Crear, LFE */
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MODE_7_1_BACK = 12, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */
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MODE_7_1_TOP_FRONT = 14, /**< C, L+R, LS+RS, LFE, Ltop+Rtop */
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MODE_7_1_REAR_SURROUND = 33, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */
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MODE_7_1_FRONT_CENTER = 34, /**< C, LC+RC, L+R, LS+RS, LFE */
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MODE_212 = 128 /**< 212 configuration, used in ELDv2 */
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} CHANNEL_MODE;
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/**
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* Speaker description tags.
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* Do not change the enumeration values unless it keeps the following
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* segmentation:
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* - Bit 0-3: Horizontal postion (0: none, 1: front, 2: side, 3: back, 4: lfe)
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* - Bit 4-7: Vertical position (0: normal, 1: top, 2: bottom)
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*/
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typedef enum {
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ACT_NONE = 0x00,
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ACT_FRONT = 0x01, /*!< Front speaker position (at normal height) */
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ACT_SIDE = 0x02, /*!< Side speaker position (at normal height) */
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ACT_BACK = 0x03, /*!< Back speaker position (at normal height) */
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ACT_LFE = 0x04, /*!< Low frequency effect speaker postion (front) */
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ACT_TOP =
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0x10, /*!< Top speaker area (for combination with speaker positions) */
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ACT_FRONT_TOP = 0x11, /*!< Top front speaker = (ACT_FRONT|ACT_TOP) */
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ACT_SIDE_TOP = 0x12, /*!< Top side speaker = (ACT_SIDE |ACT_TOP) */
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ACT_BACK_TOP = 0x13, /*!< Top back speaker = (ACT_BACK |ACT_TOP) */
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ACT_BOTTOM =
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0x20, /*!< Bottom speaker area (for combination with speaker positions) */
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ACT_FRONT_BOTTOM = 0x21, /*!< Bottom front speaker = (ACT_FRONT|ACT_BOTTOM) */
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ACT_SIDE_BOTTOM = 0x22, /*!< Bottom side speaker = (ACT_SIDE |ACT_BOTTOM) */
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ACT_BACK_BOTTOM = 0x23 /*!< Bottom back speaker = (ACT_BACK |ACT_BOTTOM) */
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} AUDIO_CHANNEL_TYPE;
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typedef enum {
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SIG_UNKNOWN = -1,
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SIG_IMPLICIT = 0,
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SIG_EXPLICIT_BW_COMPATIBLE = 1,
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SIG_EXPLICIT_HIERARCHICAL = 2
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} SBR_PS_SIGNALING;
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/**
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* Audio Codec flags.
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*/
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#define AC_ER_VCB11 \
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0x000001 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use \
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virtual codebooks */
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#define AC_ER_RVLC \
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0x000002 /*!< aacSpectralDataResilienceFlag flag (from ASC): 1 means use \
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huffman codeword reordering */
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#define AC_ER_HCR \
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0x000004 /*!< aacSectionDataResilienceFlag flag (from ASC): 1 means use \
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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_RSV603DA 0x000200 /*!< RSVD60 3D audio */
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#define AC_HDAAC 0x000400 /*!< HD-AAC */
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#define AC_RSVD50 0x004000 /*!< Rsvd50 */
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#define AC_SBR_PRESENT 0x008000 /*!< SBR present flag (from ASC) */
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#define AC_SBRCRC \
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0x010000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */
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#define AC_PS_PRESENT 0x020000 /*!< PS present flag (from ASC or implicit) */
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#define AC_MPS_PRESENT \
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0x040000 /*!< MPS present flag (from ASC or implicit) \
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*/
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#define AC_DRM 0x080000 /*!< DRM bit stream syntax */
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#define AC_INDEP 0x100000 /*!< Independency flag */
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#define AC_MPEGD_RES 0x200000 /*!< MPEG-D residual individual channel data. */
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#define AC_SAOC_PRESENT 0x400000 /*!< SAOC Present Flag */
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#define AC_DAB 0x800000 /*!< DAB bit stream syntax */
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#define AC_ELD_DOWNSCALE 0x1000000 /*!< ELD Downscaled playout */
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#define AC_LD_MPS 0x2000000 /*!< Low Delay MPS. */
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#define AC_DRC_PRESENT \
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0x4000000 /*!< Dynamic Range Control (DRC) data found. \
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*/
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#define AC_USAC_SCFGI3 \
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0x8000000 /*!< USAC flag: If stereoConfigIndex is 3 the flag is set. */
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/**
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* Audio Codec flags (reconfiguration).
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*/
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#define AC_CM_DET_CFG_CHANGE \
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0x000001 /*!< Config mode signalizes the callback to work in config change \
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detection mode */
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#define AC_CM_ALLOC_MEM \
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0x000002 /*!< Config mode signalizes the callback to work in memory \
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allocation mode */
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/**
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* Audio Codec flags (element specific).
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*/
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#define AC_EL_USAC_TW 0x000001 /*!< USAC time warped filter bank is active */
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#define AC_EL_USAC_NOISE 0x000002 /*!< USAC noise filling is active */
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#define AC_EL_USAC_ITES 0x000004 /*!< USAC SBR inter-TES tool is active */
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#define AC_EL_USAC_PVC \
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0x000008 /*!< USAC SBR predictive vector coding tool is active */
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#define AC_EL_USAC_MPS212 0x000010 /*!< USAC MPS212 tool is active */
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#define AC_EL_USAC_LFE 0x000020 /*!< USAC element is LFE */
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#define AC_EL_USAC_CP_POSSIBLE \
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0x000040 /*!< USAC may use Complex Stereo Prediction in this channel element \
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*/
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#define AC_EL_ENHANCED_NOISE 0x000080 /*!< Enhanced noise filling*/
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#define AC_EL_IGF_AFTER_TNS 0x000100 /*!< IGF after TNS */
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#define AC_EL_IGF_INDEP_TILING 0x000200 /*!< IGF independent tiling */
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#define AC_EL_IGF_USE_ENF 0x000400 /*!< IGF use enhanced noise filling */
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#define AC_EL_FULLBANDLPD 0x000800 /*!< enable fullband LPD tools */
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#define AC_EL_LPDSTEREOIDX 0x001000 /*!< LPD-stereo-tool stereo index */
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#define AC_EL_LFE 0x002000 /*!< The element is of type LFE. */
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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_SAC 0x00020000
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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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UCHAR channelConfigZero; /**< Use channel config zero + pce although a
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standard channel config could be signaled. */
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INT samplingRate; /**< Sampling rate. */
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INT extSamplingRate; /**< Extended samplerate (SBR). */
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INT downscaleSamplingRate; /**< Downscale sampling rate (ELD downscaled mode)
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*/
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INT bitRate; /**< Average bitrate. */
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int samplesPerFrame; /**< Number of PCM samples per codec frame and audio
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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
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transmitted in a super-frame (BSAC). */
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int BSAClayerLength; /**< The average length of the large-step layers in bytes
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(BSAC). */
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UINT flags; /**< flags */
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UCHAR matrixMixdownA; /**< Matrix mixdown index to put into PCE. Default value
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0 means no mixdown coefficient, valid values are 1-4
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which correspond to matrix_mixdown_idx 0-3. */
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UCHAR headerPeriod; /**< Frame period for sending in band configuration
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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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SBR_PS_SIGNALING sbrSignaling; /**< 0: implicit signaling, 1: backwards
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compatible explicit signaling, 2:
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hierarcical explicit signaling */
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UCHAR rawConfig[64]; /**< raw codec specific config as bit stream */
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int rawConfigBits; /**< Size of rawConfig in bits */
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|
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 */
|
|
|
|
/* 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 */
|