mirror of
https://github.com/mstorsjo/fdk-aac.git
synced 2024-12-12 08:26:11 +01:00
6cfabd3536
Bug: 71430241 Test: CTS DecoderTest and DecoderTestAacDrc original-Change-Id: Iaa20f749b8a04d553b20247cfe1a8930ebbabe30 Apply clang-format also on header files. original-Change-Id: I14de1ef16bbc79ec0283e745f98356a10efeb2e4 Fixes for MPEG-D DRC original-Change-Id: If1de2d74bbbac84b3f67de3b88b83f6a23b8a15c Catch unsupported tw_mdct at an early stage original-Change-Id: Ied9dd00d754162a0e3ca1ae3e6b854315d818afe Fixing PVC transition frames original-Change-Id: Ib75725abe39252806c32d71176308f2c03547a4e Move qmf bands sanity check original-Change-Id: Iab540c3013c174d9490d2ae100a4576f51d8dbc4 Initialize scaling variable original-Change-Id: I3c4087101b70e998c71c1689b122b0d7762e0f9e Add 16 qmf band configuration to getSlotNrgHQ() original-Change-Id: I49a5d30f703a1b126ff163df9656db2540df21f1 Always apply byte alignment at the end of the AudioMuxElement original-Change-Id: I42d560287506d65d4c3de8bfe3eb9a4ebeb4efc7 Setup SBR element only if no parse error exists original-Change-Id: I1915b73704bc80ab882b9173d6bec59cbd073676 Additional array index check in HCR original-Change-Id: I18cc6e501ea683b5009f1bbee26de8ddd04d8267 Fix fade-in index selection in concealment module original-Change-Id: Ibf802ed6ed8c05e9257e1f3b6d0ac1162e9b81c1 Enable explicit backward compatible parser for AAC_LD original-Change-Id: I27e9c678dcb5d40ed760a6d1e06609563d02482d Skip spatial specific config in explicit backward compatible ASC original-Change-Id: Iff7cc365561319e886090cedf30533f562ea4d6e Update flags description in decoder API original-Change-Id: I9a5b4f8da76bb652f5580cbd3ba9760425c43830 Add QMF domain reset function original-Change-Id: I4f89a8a2c0277d18103380134e4ed86996e9d8d6 DRC upgrade v2.1.0 original-Change-Id: I5731c0540139dab220094cd978ef42099fc45b74 Fix integer overflow in sqrtFixp_lookup() original-Change-Id: I429a6f0d19aa2cc957e0f181066f0ca73968c914 Fix integer overflow in invSqrtNorm2() original-Change-Id: I84de5cbf9fb3adeb611db203fe492fabf4eb6155 Fix integer overflow in GenerateRandomVector() original-Change-Id: I3118a641008bd9484d479e5b0b1ee2b5d7d44d74 Fix integer overflow in adjustTimeSlot_EldGrid() original-Change-Id: I29d503c247c5c8282349b79df940416a512fb9d5 Fix integer overflow in FDKsbrEnc_codeEnvelope() original-Change-Id: I6b34b61ebb9d525b0c651ed08de2befc1f801449 Follow-up on: Fix integer overflow in adjustTimeSlot_EldGrid() original-Change-Id: I6f8f578cc7089e5eb7c7b93e580b72ca35ad689a Fix integer overflow in get_pk_v2() original-Change-Id: I63375bed40d45867f6eeaa72b20b1f33e815938c Fix integer overflow in Syn_filt_zero() original-Change-Id: Ie0c02fdfbe03988f9d3b20d10cd9fe4c002d1279 Fix integer overflow in CFac_CalcFacSignal() original-Change-Id: Id2d767c40066c591b51768e978eb8af3b803f0c5 Fix integer overflow in FDKaacEnc_FDKaacEnc_calcPeNoAH() original-Change-Id: Idcbd0f4a51ae2550ed106aa6f3d678d1f9724841 Fix integer overflow in sbrDecoder_calculateGainVec() original-Change-Id: I7081bcbe29c5cede9821b38d93de07c7add2d507 Fix integer overflow in CLpc_SynthesisLattice() original-Change-Id: I4a95ddc18de150102352d4a1845f06094764c881 Fix integer overflow in Pred_Lt4() original-Change-Id: I4dbd012b2de7d07c3e70a47b92e3bfae8dbc750a Fix integer overflow in FDKsbrEnc_InitSbrFastTransientDetector() original-Change-Id: I788cbec1a4a00f44c2f3a72ad7a4afa219807d04 Fix unsigned integer overflow in FDKaacEnc_WriteBitstream() original-Change-Id: I68fc75166e7d2cd5cd45b18dbe3d8c2a92f1822a Fix unsigned integer overflow in FDK_MetadataEnc_Init() original-Change-Id: Ie8d025f9bcdb2442c704bd196e61065c03c10af4 Fix overflow in pseudo random number generators original-Change-Id: I3e2551ee01356297ca14e3788436ede80bd5513c Fix unsigned integer overflow in sbrDecoder_Parse() original-Change-Id: I3f231b2f437e9c37db4d5b964164686710eee971 Fix unsigned integer overflow in longsub() original-Change-Id: I73c2bc50415cac26f1f5a29e125bbe75f9180a6e Fix unsigned integer overflow in CAacDecoder_DecodeFrame() original-Change-Id: Ifce2db4b1454b46fa5f887e9d383f1cc43b291e4 Fix overflow at CLpdChannelStream_Read() original-Change-Id: Idb9d822ce3a4272e4794b643644f5434e2d4bf3f Fix unsigned integer overflow in Hcr_State_BODY_SIGN_ESC__ESC_WORD() original-Change-Id: I1ccf77c0015684b85534c5eb97162740a870b71c Fix unsigned integer overflow in UsacConfig_Parse() original-Change-Id: Ie6d27f84b6ae7eef092ecbff4447941c77864d9f Fix unsigned integer overflow in aacDecoder_drcParse() original-Change-Id: I713f28e883eea3d70b6fa56a7b8f8c22bcf66ca0 Fix unsigned integer overflow in aacDecoder_drcReadCompression() original-Change-Id: Ia34dfeb88c4705c558bce34314f584965cafcf7a Fix unsigned integer overflow in CDataStreamElement_Read() original-Change-Id: Iae896cc1d11f0a893d21be6aa90bd3e60a2c25f0 Fix unsigned integer overflow in transportDec_AdjustEndOfAccessUnit() original-Change-Id: I64cf29a153ee784bb4a16fdc088baabebc0007dc Fix unsigned integer overflow in transportDec_GetAuBitsRemaining() original-Change-Id: I975b3420faa9c16a041874ba0db82e92035962e4 Fix unsigned integer overflow in extractExtendedData() original-Change-Id: I2a59eb09e2053cfb58dfb75fcecfad6b85a80a8f Fix signed integer overflow in CAacDecoder_ExtPayloadParse() original-Change-Id: I4ad5ca4e3b83b5d964f1c2f8c5e7b17c477c7929 Fix unsigned integer overflow in CAacDecoder_DecodeFrame() original-Change-Id: I29a39df77d45c52a0c9c5c83c1ba81f8d0f25090 Follow-up on: Fix integer overflow in CLpc_SynthesisLattice() original-Change-Id: I8fb194ffc073a3432a380845be71036a272d388f Fix signed integer overflow in _interpolateDrcGain() original-Change-Id: I879ec9ab14005069a7c47faf80e8bc6e03d22e60 Fix unsigned integer overflow in FDKreadBits() original-Change-Id: I1f47a6a8037ff70375aa8844947d5681bb4287ad Fix unsigned integer overflow in FDKbyteAlign() original-Change-Id: Id5f3a11a0c9e50fc6f76ed6c572dbd4e9f2af766 Fix unsigned integer overflow in FDK_get32() original-Change-Id: I9d33b8e97e3d38cbb80629cb859266ca0acdce96 Fix unsigned integer overflow in FDK_pushBack() original-Change-Id: Ic87f899bc8c6acf7a377a8ca7f3ba74c3a1e1c19 Fix unsigned integer overflow in FDK_pushForward() original-Change-Id: I3b754382f6776a34be1602e66694ede8e0b8effc Fix unsigned integer overflow in ReadPsData() original-Change-Id: I25361664ba8139e32bbbef2ca8c106a606ce9c37 Fix signed integer overflow in E_UTIL_residu() original-Change-Id: I8c3abd1f437ee869caa8fb5903ce7d3d641b6aad REVERT: Follow-up on: Integer overflow in CLpc_SynthesisLattice(). original-Change-Id: I3d340099acb0414795c8dfbe6362bc0a8f045f9b Follow-up on: Fix integer overflow in CLpc_SynthesisLattice() original-Change-Id: I4aedb8b3a187064e9f4d985175aa55bb99cc7590 Follow-up on: Fix unsigned integer overflow in aacDecoder_drcParse() original-Change-Id: I2aa2e13916213bf52a67e8b0518e7bf7e57fb37d Fix integer overflow in acelp original-Change-Id: Ie6390c136d84055f8b728aefbe4ebef6e029dc77 Fix unsigned integer overflow in aacDecoder_UpdateBitStreamCounters() original-Change-Id: I391ffd97ddb0b2c184cba76139bfb356a3b4d2e2 Adjust concealment default settings original-Change-Id: I6a95db935a327c47df348030bcceafcb29f54b21 Saturate estimatedStartPos original-Change-Id: I27be2085e0ae83ec9501409f65e003f6bcba1ab6 Negative shift exponent in _interpolateDrcGain() original-Change-Id: I18edb26b26d002aafd5e633d4914960f7a359c29 Negative shift exponent in calculateICC() original-Change-Id: I3dcd2ae98d2eb70ee0d59750863cbb2a6f4f8aba Too large shift exponent in FDK_put() original-Change-Id: Ib7d9aaa434d2d8de4a13b720ca0464b31ca9b671 Too large shift exponent in CalcInvLdData() original-Change-Id: I43e6e78d4cd12daeb1dcd5d82d1798bdc2550262 Member access within null pointer of type SBR_CHANNEL original-Change-Id: Idc5e4ea8997810376d2f36bbdf628923b135b097 Member access within null pointer of type CpePersistentData original-Change-Id: Ib6c91cb0d37882768e5baf63324e429589de0d9d Member access within null pointer FDKaacEnc_psyMain() original-Change-Id: I7729b7f4479970531d9dc823abff63ca52e01997 Member access within null pointer FDKaacEnc_GetPnsParam() original-Change-Id: I9aa3b9f3456ae2e0f7483dbd5b3dde95fc62da39 Member access within null pointer FDKsbrEnc_EnvEncodeFrame() original-Change-Id: I67936f90ea714e90b3e81bc0dd1472cc713eb23a Add HCR sanity check original-Change-Id: I6c1d9732ebcf6af12f50b7641400752f74be39f7 Fix memory issue for HBE edge case with 8:3 SBR original-Change-Id: I11ea58a61e69fbe8bf75034b640baee3011e63e9 Additional SBR parametrization sanity check for ELD original-Change-Id: Ie26026fbfe174c2c7b3691f6218b5ce63e322140 Add MPEG-D DRC channel layout check original-Change-Id: Iea70a74f171b227cce636a9eac4ba662777a2f72 Additional out-of-bounds checks in MPEG-D DRC original-Change-Id: Ife4a8c3452c6fde8a0a09e941154a39a769777d4 Change-Id: Ic63cb2f628720f54fe9b572b0cb528e2599c624e
466 lines
17 KiB
C
466 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 - 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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/******************* MPEG transport format encoder 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;
|
|
} CSUsacConfig;
|
|
|
|
/**
|
|
* Audio configuration struct, suitable for encoder and decoder configuration.
|
|
*/
|
|
typedef struct {
|
|
/* XYZ Specific Data */
|
|
union {
|
|
CSGaSpecificConfig
|
|
m_gaSpecificConfig; /**< General audio specific configuration. */
|
|
CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
|
|
CSUsacConfig m_usacConfig; /**< USAC specific configuration */
|
|
} m_sc;
|
|
|
|
/* Common ASC parameters */
|
|
CProgramConfig m_progrConfigElement; /**< Program configuration. */
|
|
|
|
AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */
|
|
UINT m_samplingFrequency; /**< Samplerate. */
|
|
UINT m_samplesPerFrame; /**< Amount of samples per frame. */
|
|
UINT m_directMapping; /**< Document this please !! */
|
|
|
|
AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */
|
|
UINT m_extensionSamplingFrequency; /**< Samplerate */
|
|
|
|
SCHAR m_channelConfiguration; /**< Channel configuration index */
|
|
|
|
SCHAR m_epConfig; /**< Error protection index */
|
|
SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */
|
|
SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */
|
|
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 */
|
|
|
|
} CSAudioSpecificConfig;
|
|
|
|
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. */
|
|
} CSTpCallBacks;
|
|
|
|
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};
|
|
|
|
static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
|
|
UINT sf_index;
|
|
UINT tableSize = (1 << nBits) - 1;
|
|
|
|
for (sf_index = 0; sf_index < tableSize; sf_index++) {
|
|
if (SamplingRateTable[sf_index] == samplingRate) break;
|
|
}
|
|
|
|
if (sf_index > tableSize) {
|
|
return tableSize - 1;
|
|
}
|
|
|
|
return sf_index;
|
|
}
|
|
|
|
/*
|
|
* Get Channel count from channel configuration
|
|
*/
|
|
static inline int getNumberOfTotalChannels(int channelConfig) {
|
|
switch (channelConfig) {
|
|
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;
|
|
}
|
|
}
|
|
|
|
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};
|
|
return n[channelConfig];
|
|
}
|
|
|
|
#endif /* TP_DATA_H */
|