2018-02-26 20:17:00 +01:00
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/* -----------------------------------------------------------------------------
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2012-07-11 19:15:24 +02:00
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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1. INTRODUCTION
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2. COPYRIGHT LICENSE
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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modifications thereto to recipients of copies in binary form.
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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3. NO PATENT LICENSE
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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4. DISCLAIMER
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2018-02-26 20:17:00 +01:00
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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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2012-07-11 19:15:24 +02:00
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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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2018-02-26 20:17:00 +01:00
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----------------------------------------------------------------------------- */
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/******************* MPEG transport format decoder library *********************
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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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Author(s): Manuel Jander
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2012-07-11 19:15:24 +02:00
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Description: MPEG Transport data tables
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2018-02-26 20:17:00 +01:00
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*******************************************************************************/
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2012-07-11 19:15:24 +02:00
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2018-02-26 20:17:00 +01:00
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#ifndef TP_DATA_H
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#define TP_DATA_H
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#include "machine_type.h"
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#include "FDK_audio.h"
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#include "FDK_bitstream.h"
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/*
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* Configuration
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*/
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#define TP_USAC_MAX_SPEAKERS (24)
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#define TP_USAC_MAX_EXT_ELEMENTS ((24))
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#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS)
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#define TP_USAC_MAX_CONFIG_LEN \
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512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \
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AudioPreRoll() (285) */
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#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \
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(1) /* Number of frames for config change in USAC */
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enum {
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TPDEC_FLUSH_OFF = 0,
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TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
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TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
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TPDEC_USAC_DASH_IPF_FLUSH_ON = 3
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};
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enum {
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TPDEC_BUILD_UP_OFF = 0,
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TPDEC_RSV60_BUILD_UP_ON = 1,
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TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
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TPDEC_USAC_BUILD_UP_ON = 3,
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TPDEC_RSV60_BUILD_UP_IDLE = 4,
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TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
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};
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/**
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* ProgramConfig struct.
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*/
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/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
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#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
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#define PC_LFE_CHANNELS_MAX 4
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#define PC_ASSOCDATA_MAX 8
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#define PC_CCEL_MAX 16 /* CC elements */
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#define PC_COMMENTLENGTH 256
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#define PC_NUM_HEIGHT_LAYER 3
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typedef struct {
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/* PCE bitstream elements: */
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UCHAR ElementInstanceTag;
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UCHAR Profile;
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UCHAR SamplingFrequencyIndex;
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UCHAR NumFrontChannelElements;
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UCHAR NumSideChannelElements;
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UCHAR NumBackChannelElements;
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UCHAR NumLfeChannelElements;
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UCHAR NumAssocDataElements;
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UCHAR NumValidCcElements;
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UCHAR MonoMixdownPresent;
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UCHAR MonoMixdownElementNumber;
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UCHAR StereoMixdownPresent;
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UCHAR StereoMixdownElementNumber;
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UCHAR MatrixMixdownIndexPresent;
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UCHAR MatrixMixdownIndex;
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UCHAR PseudoSurroundEnable;
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UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
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UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
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UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
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UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
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UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
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UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
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UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
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UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
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UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
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UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
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UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
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UCHAR CcElementIsIndSw[PC_CCEL_MAX];
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UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
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UCHAR CommentFieldBytes;
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UCHAR Comment[PC_COMMENTLENGTH];
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/* Helper variables for administration: */
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UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */
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UCHAR
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NumChannels; /*!< Amount of audio channels summing all channel elements
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including LFEs */
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UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs
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and CPEs */
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UCHAR elCounter;
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} CProgramConfig;
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typedef enum {
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ASCEXT_UNKOWN = -1,
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ASCEXT_SBR = 0x2b7,
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ASCEXT_PS = 0x548,
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ASCEXT_MPS = 0x76a,
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ASCEXT_SAOC = 0x7cb,
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ASCEXT_LDMPS = 0x7cc
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} TP_ASC_EXTENSION_ID;
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/**
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* GaSpecificConfig struct
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*/
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typedef struct {
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UINT m_frameLengthFlag;
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UINT m_dependsOnCoreCoder;
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UINT m_coreCoderDelay;
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UINT m_extensionFlag;
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UINT m_extensionFlag3;
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UINT m_layer;
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UINT m_numOfSubFrame;
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UINT m_layerLength;
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} CSGaSpecificConfig;
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typedef enum {
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ELDEXT_TERM = 0x0, /* Termination tag */
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ELDEXT_SAOC = 0x1, /* SAOC config */
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ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */
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ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */
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/* reserved */
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} ASC_ELD_EXT_TYPE;
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typedef struct {
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UCHAR m_frameLengthFlag;
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UCHAR m_sbrPresentFlag;
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UCHAR
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m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
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UCHAR m_sbrSamplingRate;
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UCHAR m_sbrCrcFlag;
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UINT m_downscaledSamplingFrequency;
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} CSEldSpecificConfig;
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typedef struct {
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USAC_EXT_ELEMENT_TYPE usacExtElementType;
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USHORT usacExtElementConfigLength;
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USHORT usacExtElementDefaultLength;
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UCHAR usacExtElementPayloadFrag;
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UCHAR usacExtElementHasAudioPreRoll;
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} CSUsacExtElementConfig;
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typedef struct {
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MP4_ELEMENT_ID usacElementType;
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UCHAR m_noiseFilling;
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UCHAR m_harmonicSBR;
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UCHAR m_interTes;
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UCHAR m_pvc;
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UCHAR m_stereoConfigIndex;
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CSUsacExtElementConfig extElement;
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} CSUsacElementConfig;
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typedef struct {
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UCHAR m_frameLengthFlag;
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UCHAR m_coreSbrFrameLengthIndex;
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UCHAR m_sbrRatioIndex;
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UCHAR m_nUsacChannels; /* number of audio channels signaled in
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UsacDecoderConfig() / rsv603daDecoderConfig() via
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numElements and usacElementType */
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UCHAR m_channelConfigurationIndex;
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UINT m_usacNumElements;
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CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS];
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UCHAR numAudioChannels;
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UCHAR m_usacConfigExtensionPresent;
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UCHAR elementLengthPresent;
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UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN];
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USHORT UsacConfigBits;
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} CSUsacConfig;
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/**
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* Audio configuration struct, suitable for encoder and decoder configuration.
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*/
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typedef struct {
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/* XYZ Specific Data */
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union {
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CSGaSpecificConfig
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m_gaSpecificConfig; /**< General audio specific configuration. */
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CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
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CSUsacConfig m_usacConfig; /**< USAC specific configuration */
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} m_sc;
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/* Common ASC parameters */
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CProgramConfig m_progrConfigElement; /**< Program configuration. */
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AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */
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UINT m_samplingFrequency; /**< Samplerate. */
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UINT m_samplesPerFrame; /**< Amount of samples per frame. */
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UINT m_directMapping; /**< Document this please !! */
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AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */
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UINT m_extensionSamplingFrequency; /**< Samplerate */
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SCHAR m_channelConfiguration; /**< Channel configuration index */
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SCHAR m_epConfig; /**< Error protection index */
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SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */
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SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */
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|
|
SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */
|
|
|
|
|
|
|
|
SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the
|
|
|
|
bitstream */
|
|
|
|
SCHAR
|
|
|
|
m_psPresentFlag; /**< Flag indicating the presence of parametric stereo
|
|
|
|
data in the bitstream */
|
|
|
|
UCHAR m_samplingFrequencyIndex; /**< Samplerate index */
|
|
|
|
UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
|
|
|
|
SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */
|
|
|
|
|
|
|
|
UCHAR
|
|
|
|
configMode; /**< The flag indicates if the callback shall work in memory
|
|
|
|
allocation mode or in config change detection mode */
|
|
|
|
UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
|
|
|
|
parameter has changed that requires a memory
|
|
|
|
reconfiguration, otherwise it will be cleared */
|
|
|
|
UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
|
|
|
|
parameter has changed that requires a memory
|
|
|
|
reconfiguration, otherwise it will be cleared */
|
|
|
|
UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
|
|
|
|
parameter has changed that requires a memory
|
|
|
|
reconfiguration, otherwise it will be cleared */
|
|
|
|
|
|
|
|
UCHAR
|
|
|
|
config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
|
|
|
|
UINT configBits; /**< Configuration length in bits */
|
2012-07-11 19:15:24 +02:00
|
|
|
|
|
|
|
} CSAudioSpecificConfig;
|
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
typedef struct {
|
|
|
|
SCHAR flushCnt; /**< Flush frame counter */
|
|
|
|
UCHAR flushStatus; /**< Flag indicates flush mode: on|off */
|
|
|
|
SCHAR buildUpCnt; /**< Build up frame counter */
|
|
|
|
UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
|
|
|
|
UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
|
|
|
|
needs to be initialized again via callback. Make sure
|
|
|
|
that memory is freed before initialization. */
|
|
|
|
UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
|
|
|
|
right truncation occured before */
|
|
|
|
UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
|
|
|
|
even if new config is the same */
|
|
|
|
} CCtrlCFGChange;
|
|
|
|
|
|
|
|
typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
|
|
|
|
const UCHAR configMode, UCHAR *configChanged);
|
|
|
|
typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
|
|
|
|
typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
|
|
|
|
typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
|
|
|
|
const AUDIO_OBJECT_TYPE coreCodec,
|
|
|
|
const INT samplingRate, const INT stereoConfigIndex,
|
|
|
|
const INT coreSbrFrameLengthIndex, const INT configBytes,
|
|
|
|
const UCHAR configMode, UCHAR *configChanged);
|
|
|
|
|
|
|
|
typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
|
|
|
|
const INT sampleRateIn, const INT sampleRateOut,
|
|
|
|
const INT samplesPerFrame,
|
|
|
|
const AUDIO_OBJECT_TYPE coreCodec,
|
|
|
|
const MP4_ELEMENT_ID elementID, const INT elementIndex,
|
|
|
|
const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
|
|
|
|
const UCHAR configMode, UCHAR *configChanged,
|
|
|
|
const INT downscaleFactor);
|
|
|
|
|
|
|
|
typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
|
|
|
|
|
|
|
|
typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
|
|
|
|
const INT fullPayloadLength, const INT payloadType,
|
|
|
|
const INT subStreamIndex, const INT payloadStart,
|
|
|
|
const AUDIO_OBJECT_TYPE);
|
|
|
|
|
|
|
|
typedef struct {
|
|
|
|
cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
|
|
|
|
notify callback. */
|
|
|
|
void *cbUpdateConfigData; /*!< User data pointer for Config change notify
|
|
|
|
callback. */
|
|
|
|
cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */
|
|
|
|
void *cbFreeMemData; /*!< User data pointer for free memory callback. */
|
|
|
|
cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
|
|
|
|
control callback. */
|
|
|
|
void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
|
|
|
|
callback. */
|
|
|
|
cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */
|
|
|
|
void *cbSscData; /*!< User data pointer for SSC parser callback. */
|
|
|
|
cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */
|
|
|
|
void *cbSbrData; /*!< User data pointer for SBR header parser callback. */
|
|
|
|
cbUsac_t cbUsac;
|
|
|
|
void *cbUsacData;
|
|
|
|
cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
|
|
|
|
loudnessInfoSet parser callback. */
|
|
|
|
void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and
|
|
|
|
loudnessInfoSet parser callback. */
|
2012-07-11 19:15:24 +02:00
|
|
|
} CSTpCallBacks;
|
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
static const UINT SamplingRateTable[] = {
|
|
|
|
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
|
|
|
|
8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600,
|
|
|
|
20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0};
|
2012-07-11 19:15:24 +02:00
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
|
|
|
|
UINT sf_index;
|
|
|
|
UINT tableSize = (1 << nBits) - 1;
|
2012-07-11 19:15:24 +02:00
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
for (sf_index = 0; sf_index < tableSize; sf_index++) {
|
|
|
|
if (SamplingRateTable[sf_index] == samplingRate) break;
|
2012-07-11 19:15:24 +02:00
|
|
|
}
|
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
if (sf_index > tableSize) {
|
|
|
|
return tableSize - 1;
|
2012-07-11 19:15:24 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
return sf_index;
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Get Channel count from channel configuration
|
|
|
|
*/
|
2018-02-26 20:17:00 +01:00
|
|
|
static inline int getNumberOfTotalChannels(int channelConfig) {
|
2013-12-28 01:13:22 +01:00
|
|
|
switch (channelConfig) {
|
2018-02-26 20:17:00 +01:00
|
|
|
case 1:
|
|
|
|
case 2:
|
|
|
|
case 3:
|
|
|
|
case 4:
|
|
|
|
case 5:
|
|
|
|
case 6:
|
|
|
|
return channelConfig;
|
|
|
|
case 7:
|
|
|
|
case 12:
|
|
|
|
case 14:
|
|
|
|
return 8;
|
|
|
|
case 11:
|
|
|
|
return 7;
|
|
|
|
case 13:
|
|
|
|
return 24;
|
|
|
|
default:
|
|
|
|
return 0;
|
2013-12-28 01:13:22 +01:00
|
|
|
}
|
2012-07-11 19:15:24 +02:00
|
|
|
}
|
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
static inline int getNumberOfEffectiveChannels(
|
|
|
|
const int
|
|
|
|
channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
|
|
|
|
const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0};
|
2012-07-11 19:15:24 +02:00
|
|
|
return n[channelConfig];
|
|
|
|
}
|
|
|
|
|
2018-02-26 20:17:00 +01:00
|
|
|
#endif /* TP_DATA_H */
|