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https://github.com/mstorsjo/fdk-aac.git
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5eb9ed1b1f
Bug: 176246647 Test: atest android.media.cts.DecoderTestAacFormat android.media.cts.DecoderTestXheAac android.media.cts.DecoderTestAacDrc Change-Id: Ib8b6973e9c29e13b8ef33d7736be2b977928face
467 lines
17 KiB
C
467 lines
17 KiB
C
/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2020 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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/******************* MPEG transport format decoder library *********************
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Author(s): Manuel Jander
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Description: MPEG Transport data tables
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*******************************************************************************/
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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 */
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SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the
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bitstream */
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SCHAR
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m_psPresentFlag; /**< Flag indicating the presence of parametric stereo
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data in the bitstream */
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UCHAR m_samplingFrequencyIndex; /**< Samplerate index */
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UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
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SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */
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UCHAR
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configMode; /**< The flag indicates if the callback shall work in memory
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allocation mode or in config change detection mode */
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UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
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parameter has changed that requires a memory
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reconfiguration, otherwise it will be cleared */
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UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
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parameter has changed that requires a memory
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reconfiguration, otherwise it will be cleared */
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UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
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parameter has changed that requires a memory
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reconfiguration, otherwise it will be cleared */
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UCHAR
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config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
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UINT configBits; /**< Configuration length in bits */
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} CSAudioSpecificConfig;
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typedef struct {
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SCHAR flushCnt; /**< Flush frame counter */
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UCHAR flushStatus; /**< Flag indicates flush mode: on|off */
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SCHAR buildUpCnt; /**< Build up frame counter */
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UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
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UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
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needs to be initialized again via callback. Make sure
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that memory is freed before initialization. */
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UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
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right truncation occured before */
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UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
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even if new config is the same */
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} CCtrlCFGChange;
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typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
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const UCHAR configMode, UCHAR *configChanged);
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typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
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typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
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typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
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const AUDIO_OBJECT_TYPE coreCodec,
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const INT samplingRate, const INT frameSize,
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const INT numChannels, const INT stereoConfigIndex,
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const INT coreSbrFrameLengthIndex, const INT configBytes,
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const UCHAR configMode, UCHAR *configChanged);
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typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
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const INT sampleRateIn, const INT sampleRateOut,
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const INT samplesPerFrame,
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const AUDIO_OBJECT_TYPE coreCodec,
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const MP4_ELEMENT_ID elementID, const INT elementIndex,
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const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
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const UCHAR configMode, UCHAR *configChanged,
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const INT downscaleFactor);
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typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
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typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
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const INT fullPayloadLength, const INT payloadType,
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const INT subStreamIndex, const INT payloadStart,
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const AUDIO_OBJECT_TYPE);
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typedef struct {
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cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
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notify callback. */
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void *cbUpdateConfigData; /*!< User data pointer for Config change notify
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callback. */
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cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */
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void *cbFreeMemData; /*!< User data pointer for free memory callback. */
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cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
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control callback. */
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void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
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callback. */
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cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */
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void *cbSscData; /*!< User data pointer for SSC parser callback. */
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cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */
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void *cbSbrData; /*!< User data pointer for SBR header parser callback. */
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cbUsac_t cbUsac;
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void *cbUsacData;
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cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
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loudnessInfoSet parser callback. */
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void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and
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loudnessInfoSet parser callback. */
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} CSTpCallBacks;
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static const UINT SamplingRateTable[] = {
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96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
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8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600,
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20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0};
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static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
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UINT sf_index;
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UINT tableSize = (1 << nBits) - 1;
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for (sf_index = 0; sf_index < tableSize; sf_index++) {
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if (SamplingRateTable[sf_index] == samplingRate) break;
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}
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if (sf_index > tableSize) {
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return tableSize - 1;
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}
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return sf_index;
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}
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/*
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* Get Channel count from channel configuration
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*/
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static inline int getNumberOfTotalChannels(int channelConfig) {
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switch (channelConfig) {
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case 1:
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case 2:
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case 3:
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case 4:
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case 5:
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case 6:
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return channelConfig;
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case 7:
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case 12:
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case 14:
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return 8;
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case 11:
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return 7;
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case 13:
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return 24;
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default:
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return 0;
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}
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}
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static inline int getNumberOfEffectiveChannels(
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const int
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channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
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const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0};
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return n[channelConfig];
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}
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#endif /* TP_DATA_H */
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